From 178ff7c6f3916aff3c3eaaec8636be3b41e93011 Mon Sep 17 00:00:00 2001 From: John Lin Date: Mon, 15 Feb 2016 10:40:17 +0800 Subject: ASoC: rt5645: Add dmi_system_id "Google Setzer" Add platform specific data for Setzer project. Signed-off-by: John Lin Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7af5e73..dff706a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3557,6 +3557,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Setzer", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"), + }, + }, { } }; -- cgit v0.10.2 From a686632fd9a857776798f3479e2b58b07d938076 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 12:18:33 +0100 Subject: ALSA: hda - Split out Intel-specific codes from patch_generic_hdmi() We have too many Intel-specific codes in patch_hdmi_generic() despite its function name. And this makes it difficult to adjust per chipset, e.g. for allowing the audio notifier on an old chipset, one would need to add an explicit if() check. This patch attempts some code refactoring and cleanups in this regard; the Intel-specific codes are moved out of patch_generic_hdmi() into the new functions, patch_i915_hsw_hdmi() and patch_i915_byt_hdmi(), depending on the chipset. The other old Intel chipsets keep using patch_generic_hdmi() without Intel hacks. The existing patch_generic_hdmi() is also split to a few components so that they can be called from the Intel codec parsers. There are still many is_haswell*() and is_valleyview*() macro usages in the code. They will be cleaned up later. For the time being, only the entry are concerned. Along with this change, the i915_bound flag and the on-demand i915 component binding have been removed as a cleanup, since there is no user at this moment. This will be added back later once when Cougar Point and else start using the i915 eld notifier. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..48c63fe 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -154,7 +154,6 @@ struct hdmi_spec { /* i915/powerwell (Haswell+/Valleyview+) specific */ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ struct i915_audio_component_audio_ops i915_audio_ops; - bool i915_bound; /* was i915 bound in this driver? */ struct hdac_chmap chmap; }; @@ -2074,6 +2073,18 @@ static void hdmi_array_free(struct hdmi_spec *spec) snd_array_free(&spec->cvts); } +static void generic_spec_free(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec) { + hdmi_array_free(spec); + kfree(spec); + codec->spec = NULL; + } + codec->dp_mst = false; +} + static void generic_hdmi_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2098,10 +2109,7 @@ static void generic_hdmi_free(struct hda_codec *codec) spec->pcm_rec[pcm_idx].jack = NULL; } - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - hdmi_array_free(spec); - kfree(spec); + generic_spec_free(codec); } #ifdef CONFIG_PM @@ -2139,6 +2147,54 @@ static const struct hdmi_ops generic_standard_hdmi_ops = { .setup_stream = hdmi_setup_stream, }; +/* allocate codec->spec and assign/initialize generic parser ops */ +static int alloc_generic_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + spec->ops = generic_standard_hdmi_ops; + mutex_init(&spec->pcm_lock); + snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); + + spec->chmap.ops.get_chmap = hdmi_get_chmap; + spec->chmap.ops.set_chmap = hdmi_set_chmap; + spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + + codec->spec = spec; + hdmi_array_init(spec, 4); + + codec->patch_ops = generic_hdmi_patch_ops; + + return 0; +} + +/* generic HDMI parser */ +static int patch_generic_hdmi(struct hda_codec *codec) +{ + int err; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; + } + + generic_hdmi_init_per_pins(codec); + return 0; +} + +/* + * Intel codec parsers and helpers + */ + static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t nid) { @@ -2234,92 +2290,96 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) check_presence_and_report(codec, pin_nid); } -static int patch_generic_hdmi(struct hda_codec *codec) +/* register i915 component pin_eld_notify callback */ +static void register_i915_notifier(struct hda_codec *codec) { - struct hdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - spec->ops = generic_standard_hdmi_ops; - mutex_init(&spec->pcm_lock); - snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); + struct hdmi_spec *spec = codec->spec; - spec->chmap.ops.get_chmap = hdmi_get_chmap; - spec->chmap.ops.set_chmap = hdmi_set_chmap; - spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->use_acomp_notifier = true; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); +} - codec->spec = spec; - hdmi_array_init(spec, 4); +/* Intel Haswell and onwards; audio component with eld notifier */ +static int patch_i915_hsw_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; -#ifdef CONFIG_SND_HDA_I915 - /* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */ - if ((codec->core.vendor_id >> 16) == 0x8086 && - is_haswell_plus(codec)) { -#if 0 - /* on-demand binding leads to an unbalanced refcount when - * both i915 and hda drivers are probed concurrently; - * disabled temporarily for now - */ - if (!codec->bus->core.audio_component) - if (!snd_hdac_i915_init(&codec->bus->core)) - spec->i915_bound = true; -#endif - /* use i915 audio component notifier for hotplug */ - if (codec->bus->core.audio_component) - spec->use_acomp_notifier = true; + /* HSW+ requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } -#endif - if (is_haswell_plus(codec)) { - intel_haswell_enable_all_pins(codec, true); - intel_haswell_fixup_enable_dp12(codec); - } + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - /* For Valleyview/Cherryview, only the display codec is in the display - * power well and can use link_power ops to request/release the power. - * For Haswell/Broadwell, the controller is also in the power well and + intel_haswell_enable_all_pins(codec, true); + intel_haswell_fixup_enable_dp12(codec); + + /* For Haswell/Broadwell, the controller is also in the power well and * can cover the codec power request, and so need not set this flag. - * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec) || - is_broxton(codec)) + if (!is_haswell(codec) && !is_broadwell(codec)) codec->core.link_power_control = 1; - if (hdmi_parse_codec(codec) < 0) { - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - codec->spec = NULL; - kfree(spec); - return -EINVAL; + codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - codec->patch_ops = generic_hdmi_patch_ops; - if (is_haswell_plus(codec)) { - codec->patch_ops.set_power_state = haswell_set_power_state; - codec->dp_mst = true; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; +} + +/* Intel Baytrail and Braswell; without get_eld notifier */ +static int patch_i915_byt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } - /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - codec->auto_runtime_pm = 1; + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - generic_hdmi_init_per_pins(codec); + /* For Valleyview/Cherryview, only the display codec is in the display + * power well and can use link_power ops to request/release the power. + */ + codec->core.link_power_control = 1; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); + generic_hdmi_init_per_pins(codec); return 0; } @@ -3498,14 +3558,14 @@ HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), +HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), -- cgit v0.10.2 From 44bb6d0c3f690e60670517859925417bd7c42a22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 12:36:44 +0100 Subject: ALSA: hda - Apply AMP fix in hdmi_setup_audio_infoframe() generically The need for reprogramming the AMP mute bit at each audio info frame setup isn't always specific to Intel chips. It's safer to set it generically for all codecs with the amp bit, as this verb execution itself isn't too much load. This eliminates one usage of is_haswell_plus() macro, after all. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 48c63fe..fcd207d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -683,7 +683,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; - if (is_haswell_plus(codec)) + /* some HW (e.g. HSW+) needs reprogramming the amp at each time */ + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); -- cgit v0.10.2 From 2c1c9b86c6b22dc0cbac3f4ca2c8272c472dc463 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 12:42:06 +0100 Subject: ALSA: hda - Override HDMI setup_stream ops for Intel HSW+ Instead of checking at each time with is_haswell_plus() macro, override the setup_stream ops itself for HSW+ chips. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fcd207d..9855188 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -864,9 +864,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, struct hdmi_spec *spec = codec->spec; int err; - if (is_haswell_plus(codec)) - haswell_verify_D0(codec, cvt_nid, pin_nid); - err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); if (err) { @@ -2307,6 +2304,14 @@ static void register_i915_notifier(struct hda_codec *codec) snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } +/* setup_stream ops override for HSW+ */ +static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format) +{ + haswell_verify_D0(codec, cvt_nid, pin_nid); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); +} + /* Intel Haswell and onwards; audio component with eld notifier */ static int patch_i915_hsw_hdmi(struct hda_codec *codec) { @@ -2338,6 +2343,8 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) codec->depop_delay = 0; codec->auto_runtime_pm = 1; + spec->ops.setup_stream = i915_hsw_setup_stream; + err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); -- cgit v0.10.2 From 4846a67eb5a1d7cac76e1b22f66e88a8cbbdff3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 12:56:46 +0100 Subject: ALSA: hda - Introduce pin_cvt_fixup() ops to hdmi parser For reducing the splat of is_haswell_plus() or such macros, this patch introduces pin_cvt_fixup() ops to hdmi_spec. For HSW+ and VLV+ codecs, set this ops so that the driver can call the Intel-specific workarounds appropriately. A gratis bonus that we can remove the mux_id argument from hdmi_choose_cvt(), too, since the fixup function always refers the mux_idx from the given per_pin object. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9855188..3481b43 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -114,6 +114,9 @@ struct hdmi_ops { int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format); + void (*pin_cvt_fixup)(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid); }; struct hdmi_pcm { @@ -881,7 +884,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, * of the pin. */ static int hdmi_choose_cvt(struct hda_codec *codec, - int pin_idx, int *cvt_id, int *mux_id) + int pin_idx, int *cvt_id) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; @@ -922,8 +925,6 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_id) *cvt_id = cvt_idx; - if (mux_id) - *mux_id = mux_idx; return 0; } @@ -1016,9 +1017,6 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, int mux_idx; struct hdmi_spec *spec = codec->spec; - if (!is_haswell_plus(codec) && !is_valleyview_plus(codec)) - return; - /* On Intel platform, the mapping of converter nid to * mux index of the pins are always the same. * The pin nid may be 0, this means all pins will not @@ -1029,6 +1027,17 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, intel_not_share_assigned_cvt(codec, pin_nid, mux_idx); } +/* skeleton caller of pin_cvt_fixup ops */ +static void pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->ops.pin_cvt_fixup) + spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid); +} + /* called in hdmi_pcm_open when no pin is assigned to the PCM * in dyn_pcm_assign mode. */ @@ -1046,7 +1055,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, if (pcm_idx < 0) return -EINVAL; - err = hdmi_choose_cvt(codec, -1, &cvt_idx, NULL); + err = hdmi_choose_cvt(codec, -1, &cvt_idx); if (err) return err; @@ -1054,7 +1063,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - intel_not_share_assigned_cvt_nid(codec, 0, per_cvt->cvt_nid); + pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid); set_bit(pcm_idx, &spec->pcm_in_use); /* todo: setup spdif ctls assign */ @@ -1086,7 +1095,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, { struct hdmi_spec *spec = codec->spec; struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, pcm_idx, mux_idx = 0; + int pin_idx, cvt_idx, pcm_idx; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; @@ -1115,7 +1124,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, } } - err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); if (err < 0) { mutex_unlock(&spec->pcm_lock); return err; @@ -1132,11 +1141,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + per_pin->mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); + pin_cvt_fixup(codec, per_pin, 0); snd_hda_spdif_ctls_assign(codec, pcm_idx, per_cvt->cvt_nid); @@ -1369,12 +1377,7 @@ static void update_eld(struct hda_codec *codec, * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, - per_pin->mux_idx); - } - + pin_cvt_fixup(codec, per_pin, 0); hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } @@ -1709,7 +1712,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, * skip pin setup and return 0 to make audio playback * be ongoing */ - intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); mutex_unlock(&spec->pcm_lock); @@ -1722,18 +1725,16 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - /* Verify pin:cvt selections to avoid silent audio after S3. - * After S3, the audio driver restores pin:cvt selections - * but this can happen before gfx is ready and such selection - * is overlooked by HW. Thus multiple pins can share a same - * default convertor and mute control will affect each other, - * which can cause a resumed audio playback become silent - * after S3. - */ - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); - } + + /* Verify pin:cvt selections to avoid silent audio after S3. + * After S3, the audio driver restores pin:cvt selections + * but this can happen before gfx is ready and such selection + * is overlooked by HW. Thus multiple pins can share a same + * default convertor and mute control will affect each other, + * which can cause a resumed audio playback become silent + * after S3. + */ + pin_cvt_fixup(codec, per_pin, 0); /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ @@ -2312,6 +2313,20 @@ static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } +/* pin_cvt_fixup ops override for HSW+ and VLV+ */ +static void i915_pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + if (per_pin) { + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, per_pin->pin_nid, + per_pin->mux_idx); + } else { + intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + } +} + /* Intel Haswell and onwards; audio component with eld notifier */ static int patch_i915_hsw_hdmi(struct hda_codec *codec) { @@ -2344,6 +2359,7 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) codec->auto_runtime_pm = 1; spec->ops.setup_stream = i915_hsw_setup_stream; + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; err = hdmi_parse_codec(codec); if (err < 0) { @@ -2381,6 +2397,8 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) codec->depop_delay = 0; codec->auto_runtime_pm = 1; + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); -- cgit v0.10.2 From e85015a3797f2665cc6f0339e6407adc00ac4245 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 13:56:19 +0100 Subject: ALSA: hda - Use eld notifier for Intel SandyBridge and IvyBridge HDMI/DP Intel SandyBridge and IvyBridge (CougarPoint and PantherPoint platforms) have also the same digital port vs audio widget mapping (from port B = NID 0x05 to port D = NID 0x07) as Haswell & co. So, we can reuse the existing functions for HSW+ for these platforms without changing there, but just by re-adding the on-demand i915 binding in HDMI codec driver. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3481b43..09eb26c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -157,6 +157,7 @@ struct hdmi_spec { /* i915/powerwell (Haswell+/Valleyview+) specific */ bool use_acomp_notifier; /* use i915 eld_notify callback for hotplug */ struct i915_audio_component_audio_ops i915_audio_ops; + bool i915_bound; /* was i915 bound in this driver? */ struct hdac_chmap chmap; }; @@ -2077,6 +2078,8 @@ static void generic_spec_free(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; if (spec) { + if (spec->i915_bound) + snd_hdac_i915_exit(&codec->bus->core); hdmi_array_free(spec); kfree(spec); codec->spec = NULL; @@ -2409,6 +2412,40 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) return 0; } +/* Intel SandyBridge and IvyBridge; with i915 eld notifier */ +static int patch_i915_cpt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* no i915 component should have been bound before this */ + if (WARN_ON(codec->bus->core.audio_component)) + return -EBUSY; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; + + /* Try to bind with i915 now */ + err = snd_hdac_i915_init(&codec->bus->core); + if (err < 0) + goto error; + spec->i915_bound = true; + + err = hdmi_parse_codec(codec); + if (err < 0) + goto error; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; + + error: + generic_spec_free(codec); + return err; +} + /* * Shared non-generic implementations */ @@ -3582,8 +3619,8 @@ HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), -- cgit v0.10.2 From d745f5e7b8b2961f68b0b9093a0f914a8a83c2ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 14:41:58 +0100 Subject: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV Intel IronLake and ValleyView platforms have different HDMI widget pin and digital port mapping from other newer ones. The recent ones (HSW+) have NID 0x05 to 0x07 for port B to port D, while these chips have NID 0x04 to 0x06. For adapting this mapping, pass the codec object instead of the bus object to snd_hdac_sync_audio_rate() and snd_hdac_acomp_get_eld() so that they can check the codec ID and calculate the mapping properly. The changes in the HDMI codec driver side will follow in the later patch. Signed-off-by: Takashi Iwai diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index fa341fc..eed87a7 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -10,8 +10,8 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); int snd_hdac_get_display_clk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate); -int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); @@ -29,12 +29,12 @@ static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) { return 0; } -static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, - int rate) +static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, + hda_nid_t nid, int rate) { return 0; } -static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, +static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index fb96aea..750a4ea 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -118,22 +118,40 @@ int snd_hdac_get_display_clk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); -/* There is a fixed mapping between audio pin node and display port - * on current Intel platforms: +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: * Pin Widget 5 - PORT B (port = 1 in i915 driver) * Pin Widget 6 - PORT C (port = 2 in i915 driver) * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) */ -static int pin2port(hda_nid_t pin_nid) +static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { - if (WARN_ON(pin_nid < 5 || pin_nid > 7)) + int base_nid; + + switch (codec->vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + base_nid = 3; + break; + default: + base_nid = 4; + break; + } + + if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) return -1; - return pin_nid - 4; + return pin_nid - base_nid; } /** * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @rate: the sample rate to set * @@ -143,14 +161,15 @@ static int pin2port(hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->sync_audio_rate(acomp->dev, port, rate); @@ -159,7 +178,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); /** * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @audio_enabled: the pointer to store the current audio state * @buffer: the buffer pointer to store ELD bytes @@ -177,16 +196,17 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->get_eld(acomp->dev, port, audio_enabled, @@ -286,6 +306,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct i915_audio_component *acomp; int ret; + if (WARN_ON(hdac_acomp)) + return -EBUSY; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 09eb26c..7e09f5e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1486,7 +1486,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1740,7 +1740,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); -- cgit v0.10.2 From 7ff652ffc06afc7f81839fb1780c57470ac09db6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Mar 2016 14:50:24 +0100 Subject: ALSA: hda - Enable i915 ELD notifier for Intel IronLake and Baytrail Since we have the fixed pin-port mapping for Intel IronLake (IbexPeak) and Baytrail (ValleyView) platforms in the code side, now it's time to add the support in the codec driver side. This patch simply enables the i915 ELD notifier for these in addition with the fix of the mapping from the port to NID in the callback. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7e09f5e..4833c7b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2274,12 +2274,23 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, static void intel_pin_eld_notify(void *audio_ptr, int port) { struct hda_codec *codec = audio_ptr; - int pin_nid = port + 0x04; + int pin_nid; /* we assume only from port-B to port-D */ if (port < 1 || port > 3) return; + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + pin_nid = port + 0x03; + break; + default: + pin_nid = port + 0x04; + break; + } + /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ @@ -2375,7 +2386,7 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec) return 0; } -/* Intel Baytrail and Braswell; without get_eld notifier */ +/* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -2409,10 +2420,11 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) } generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); return 0; } -/* Intel SandyBridge and IvyBridge; with i915 eld notifier */ +/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ static int patch_i915_cpt_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3614,11 +3626,11 @@ HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi), -- cgit v0.10.2 From 093dd449782737b50f8e1ee1608720dfd46d8ed2 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 27 Mar 2016 16:09:06 +0900 Subject: ALSA: bebob: remove needless argument from local function The 'vendor_id' argument is not used in the local function. Let's remove it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 3e4e075..932901d 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -67,7 +67,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define MODEL_MAUDIO_PROJECTMIX 0x00010091 static int -name_device(struct snd_bebob *bebob, unsigned int vendor_id) +name_device(struct snd_bebob *bebob) { struct fw_device *fw_dev = fw_parent_device(bebob->unit); char vendor[24] = {0}; @@ -232,7 +232,7 @@ bebob_probe(struct fw_unit *unit, spin_lock_init(&bebob->lock); init_waitqueue_head(&bebob->hwdep_wait); - err = name_device(bebob, entry->vendor_id); + err = name_device(bebob); if (err < 0) goto error; -- cgit v0.10.2 From 329fec2f7f7ead9dcab0a08684c700a5c55f3884 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 27 Mar 2016 16:09:07 +0900 Subject: ALSA: oxfw: remove needless member from private structure In former commit, 'struct device_info' is obsoleted, whereas private structure still keeps a pointer to it. This commit remove the member. d6ce6bbd7d83('ALSA: oxfw: rename a structure so that it means backward compatibility to old drivers') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 9beecc2..2c84714e 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -36,7 +36,6 @@ struct snd_oxfw { struct snd_card *card; struct fw_unit *unit; - const struct device_info *device_info; struct mutex mutex; spinlock_t lock; -- cgit v0.10.2 From 0655ac2f4089321766bb9af19586900ad6cef7bc Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 27 Mar 2016 16:09:08 +0900 Subject: ALSA: fireworks: move model quirk detection code to information parser Currently, model-specific quirks are detected out of information parser, however it's natural to detect it in the parser. This commit applies the idea. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8f27b67..8380fb5 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -168,6 +168,17 @@ get_hardware_info(struct snd_efw *efw) sizeof(struct snd_efw_phys_grp) * hwinfo->phys_in_grp_count); memcpy(&efw->phys_out_grps, hwinfo->phys_out_grps, sizeof(struct snd_efw_phys_grp) * hwinfo->phys_out_grp_count); + + /* AudioFire8 (since 2009) and AudioFirePre8 */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_9) + efw->is_af9 = true; + /* These models uses the same firmware. */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_2 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_4 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_9 || + hwinfo->type == MODEL_GIBSON_RIP || + hwinfo->type == MODEL_GIBSON_GOLDTOP) + efw->is_fireworks3 = true; end: kfree(hwinfo); return err; @@ -248,16 +259,6 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; - /* AudioFire8 (since 2009) and AudioFirePre8 */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) - efw->is_af9 = true; - /* These models uses the same firmware. */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_4 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_9 || - entry->model_id == MODEL_GIBSON_RIP || - entry->model_id == MODEL_GIBSON_GOLDTOP) - efw->is_fireworks3 = true; snd_efw_proc_init(efw); -- cgit v0.10.2 From 25c0e953eb56168c05fed88fbad00f76e105e2c8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 27 Mar 2016 16:09:09 +0900 Subject: ALSA: firewire-tascam: add Kconfig entry for TASCAM FW-1804 I forgot it. Fixes: 3e78e1518e12('ALSA: firewire-tascam: add support for FW-1804') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 2a779c2..ab894ed 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -134,6 +134,7 @@ config SND_FIREWIRE_TASCAM Say Y here to include support for TASCAM. * FW-1884 * FW-1082 + * FW-1804 To compile this driver as a module, choose M here: the module will be called snd-firewire-tascam. -- cgit v0.10.2 From faafd03d23c913633d2ef7e6ffebdce01b164409 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Mar 2016 12:29:24 +0200 Subject: ALSA: hda - Clear the leftover component assignment at snd_hdac_i915_exit() The commit [d745f5e7b8b2: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV] introduced a WARN_ON() to check the pointer for avoiding the double initializations. But hdac_acomp pointer wasn't cleared at snd_hdac_i915_exit(), thus after reloading the HD-audio driver, it may result in the false positive warning. This patch makes sure to clear the leftover pointer at exit. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Reported-by: Daniela Doras-prodan Signed-off-by: Takashi Iwai diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 750a4ea..ae0f305 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -372,6 +372,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } -- cgit v0.10.2 From 97cc2ed27e5a168cf423f67c3bc7c6cc41d12f82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Mar 2016 18:48:07 +0200 Subject: ALSA: hda - Fix yet another i915 pointer leftover in error path The hdac_acomp object in hdac_i915.c is left as assigned even after binding with i915 actually fails, and this leads to the WARN_ON() at the next load of the module. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=94736 Signed-off-by: Takashi Iwai diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index ae0f305..d0da250 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -339,6 +339,7 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; -- cgit v0.10.2 From 612047f0baefe2aeef1bc5ad8c7107a532b7d957 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 28 Mar 2016 14:29:22 +0100 Subject: ASoC: wm_adsp: Fix some subtle races on compressed stream Firstly, we should be locking the pwr_lock when we initialise the compressed buffer. Secondly, fixup a couple of places when we should be pulling pointers only under the pwr_lock as they may be affected by operations that take that lock. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb1..4839d19 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2240,9 +2240,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + mutex_lock(&dsp->pwr_lock); + if (wm_adsp_fw[dsp->fw].num_caps != 0) ret = wm_adsp_buffer_init(dsp); + mutex_unlock(&dsp->pwr_lock); + break; case SND_SOC_DAPM_PRE_PMD: @@ -2814,12 +2818,15 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) { - struct wm_adsp_compr_buf *buf = dsp->buffer; - struct wm_adsp_compr *compr = dsp->compr; + struct wm_adsp_compr_buf *buf; + struct wm_adsp_compr *compr; int ret = 0; mutex_lock(&dsp->pwr_lock); + buf = dsp->buffer; + compr = dsp->compr; + if (!buf) { ret = -ENODEV; goto out; @@ -2879,14 +2886,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { struct wm_adsp_compr *compr = stream->runtime->private_data; - struct wm_adsp_compr_buf *buf = compr->buf; struct wm_adsp *dsp = compr->dsp; + struct wm_adsp_compr_buf *buf; int ret = 0; adsp_dbg(dsp, "Pointer request\n"); mutex_lock(&dsp->pwr_lock); + buf = compr->buf; + if (!compr->buf) { ret = -ENXIO; goto out; -- cgit v0.10.2 From 33d740e07d1f565e44d35e7f7756a619b4f1e4ba Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 28 Mar 2016 14:29:21 +0100 Subject: ASoC: wm_adsp: Show avail in bytes to match other messages All other debug messages talk about data on the compressed stream in bytes except avail which is shown in words. To avoid confusion show avail in bytes as well. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4839d19..953c427 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2809,7 +2809,7 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) avail += wm_adsp_buffer_size(buf); adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail); + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; -- cgit v0.10.2 From 9abe3dc77ea7ccad1c2112257bb352435dcee0ff Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 28 Mar 2016 14:29:23 +0100 Subject: ASoC: cs47l24: Fix a couple of small whitespace errors Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 576087b..383700a 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1157,6 +1157,7 @@ static struct snd_compr_ops cs47l24_compr_ops = { static struct snd_soc_platform_driver cs47l24_compr_platform = { .compr_ops = &cs47l24_compr_ops, }; + static int cs47l24_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); @@ -1225,9 +1226,9 @@ static int cs47l24_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); return ret; } + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); - if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); snd_soc_unregister_platform(&pdev->dev); -- cgit v0.10.2 From c13202f7d7101a6f5542f3a31b9a6787ae7b746c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 28 Mar 2016 14:29:24 +0100 Subject: ASoC: cs47l24: Add support for audio trace firmware cs47l24 supports the audio trace firmware, this streams of audio to be captured from the CODEC over a compressed audio channel for analysis/debugging of audio processing firmwares. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 383700a..6b8b557 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "IN2L PGA", NULL, "IN2L" }, { "IN2R PGA", NULL, "IN2R" }, + { "Audio Trace DSP", NULL, "DSP2" }, + { "Audio Trace DSP", NULL, "SYSCLK" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), @@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = { .formats = CS47L24_FORMATS, }, }, + { + .name = "cs47l24-cpu-trace", + .capture = { + .stream_name = "Audio Trace CPU", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l24-dsp-trace", + .capture = { + .stream_name = "Audio Trace DSP", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + }, }; static int cs47l24_open(struct snd_compr_stream *stream) @@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream) if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { n_adsp = 2; + } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) { + n_adsp = 1; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", @@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) { struct cs47l24_priv *priv = data; struct arizona *arizona = priv->core.arizona; - int ret; + int serviced = 0; + int i, ret; + + for (i = 1; i <= 2; ++i) { + ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]); + if (ret != -ENODEV) + serviced++; + } - ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]); - if (ret == -ENODEV) { + if (!serviced) { dev_err(arizona->dev, "Spurious compressed data IRQ\n"); return IRQ_NONE; } -- cgit v0.10.2 From f17131a93f43665a76ae1f6aebdbb3e41674137c Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 29 Mar 2016 09:45:00 -0700 Subject: ASoC: intel: add function stub when ACPI is not enabled Add function stub for "sst_acpi_find_name_from_hid()" when CONFIG_ACPI is not enabled so that the driver will build successfully. This fixes the following build errors: (loadable module) ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! (or built-in) bytcr_rt5640.c:(.text+0x26fc52): undefined reference to `sst_acpi_find_name_from_hid' Reported-by: Borislav Petkov Signed-off-by: Randy Dunlap Signed-off-by: Mark Brown diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 4dcfb7e..8398cb2 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include +#include #include /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); -- cgit v0.10.2 From 92eb4f62cbac0211e43ee4a6715ee2ea43167e88 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Fri, 11 Mar 2016 10:12:56 +0530 Subject: ASoC: Intel: Bxtn: Add Broxton DSP support Broxton DSP is mostly similar to Skylake one but with subtle differences like no Code Load DMA and uses HDA DMA for code loading, DSP D0 and D3 sequences are different. These changes are comprehended by adding different DSP power up and down handlers, and new loader ops and also adding prepare and trigger which HDA DSP DMA requires Signed-off-by: Jeeja KP Signed-off-by: Jayachandran B Signed-off-by: GuruprasadX Pawse Signed-off-by: Kranthi G Signed-off-by: Dharageswari R Signed-off-by: Ramesh Babu Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c23..5a94f74 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -162,6 +162,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE + select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_HDA_I915 select SND_SOC_INTEL_SST diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index 914b6da..c28f5d0 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o + skl-sst.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c new file mode 100644 index 0000000..965ce40 --- /dev/null +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -0,0 +1,328 @@ +/* + * bxt-sst.c - DSP library functions for BXT platform + * + * Copyright (C) 2015-16 Intel Corp + * Author:Rafal Redzimski + * Jeeja KP + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include + +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-sst-ipc.h" + +#define BXT_BASEFW_TIMEOUT 3000 +#define BXT_INIT_TIMEOUT 500 +#define BXT_IPC_PURGE_FW 0x01004000 + +#define BXT_ROM_INIT 0x5 +#define BXT_ADSP_SRAM0_BASE 0x80000 + +/* Firmware status window */ +#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE +#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4) + +#define BXT_ADSP_SRAM1_BASE 0xA0000 + +static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) +{ + return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); +} + +static int sst_bxt_prepare_fw(struct sst_dsp *ctx, + const void *fwdata, u32 fwsize) +{ + int stream_tag, ret, i; + u32 reg; + + stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab); + if (stream_tag < 0) { + dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n", + stream_tag); + return stream_tag; + } + + ctx->dsp_ops.stream_tag = stream_tag; + memcpy(ctx->dmab.area, fwdata, fwsize); + + /* Purge FW request */ + sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY | + BXT_IPC_PURGE_FW | (stream_tag - 1)); + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret); + ret = -EIO; + goto base_fw_load_failed; + } + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE); + + if (reg & SKL_ADSP_REG_HIPCIE_DONE) { + sst_dsp_shim_update_bits_forced(ctx, + SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + break; + } + mdelay(1); + } + if (!i) { + dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + } + + /* enable Interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + if (SKL_FW_INIT == + (sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) & + SKL_FW_STS_MASK)) { + + dev_info(ctx->dev, "ROM loaded, continue FW loading\n"); + break; + } + mdelay(1); + } + if (!i) { + dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg); + ret = -EIO; + goto base_fw_load_failed; + } + + return ret; + +base_fw_load_failed: + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); + skl_dsp_disable_core(ctx); + return ret; +} + +static int sst_transfer_fw_host_dma(struct sst_dsp *ctx) +{ + int ret; + + ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag); + ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK, + BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot"); + + ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag); + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag); + + return ret; +} + +static int bxt_load_base_firmware(struct sst_dsp *ctx) +{ + const struct firmware *fw = NULL; + struct skl_sst *skl = ctx->thread_context; + int ret; + + ret = request_firmware(&fw, ctx->fw_name, ctx->dev); + if (ret < 0) { + dev_err(ctx->dev, "Request firmware failed %d\n", ret); + goto sst_load_base_firmware_failed; + } + + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + /* Retry Enabling core and ROM load. Retry seemed to help */ + if (ret < 0) { + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + if (ret < 0) { + dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret); + goto sst_load_base_firmware_failed; + } + } + + ret = sst_transfer_fw_host_dma(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Transfer firmware failed %d\n", ret); + dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + + skl_dsp_disable_core(ctx); + } else { + dev_dbg(ctx->dev, "Firmware download successful\n"); + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n"); + skl_dsp_disable_core(ctx); + ret = -EIO; + } else { + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + ret = 0; + } + } + +sst_load_base_firmware_failed: + release_firmware(fw); + return ret; +} + +static int bxt_set_dsp_D0(struct sst_dsp *ctx) +{ + struct skl_sst *skl = ctx->thread_context; + int ret; + + skl->boot_complete = false; + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret); + return ret; + } + + /* enable interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "ipc: error DSP boot timeout\n"); + dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + return -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + return 0; +} + +static int bxt_set_dsp_D3(struct sst_dsp *ctx) +{ + struct skl_ipc_dxstate_info dx; + struct skl_sst *skl = ctx->thread_context; + int ret = 0; + + if (!is_skl_dsp_running(ctx)) + return ret; + + dx.core_mask = SKL_DSP_CORE0_MASK; + dx.dx_mask = SKL_IPC_D3_MASK; + + ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, + SKL_BASE_FW_MODULE_ID, &dx); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret); + return ret; + } + + ret = skl_dsp_disable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret); + ret = -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RESET); + return 0; +} + +static struct skl_dsp_fw_ops bxt_fw_ops = { + .set_state_D0 = bxt_set_dsp_D0, + .set_state_D3 = bxt_set_dsp_D3, + .load_fw = bxt_load_base_firmware, + .get_fw_errcode = bxt_get_errorcode, +}; + +static struct sst_ops skl_ops = { + .irq_handler = skl_dsp_sst_interrupt, + .write = sst_shim32_write, + .read = sst_shim32_read, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .free = skl_dsp_free, +}; + +static struct sst_dsp_device skl_dev = { + .thread = skl_dsp_irq_thread_handler, + .ops = &skl_ops, +}; + +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp) +{ + struct skl_sst *skl; + struct sst_dsp *sst; + int ret; + + skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL); + if (skl == NULL) + return -ENOMEM; + + skl->dev = dev; + skl_dev.thread_context = skl; + + skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq); + if (!skl->dsp) { + dev_err(skl->dev, "skl_dsp_ctx_init failed\n"); + return -ENODEV; + } + + sst = skl->dsp; + sst->fw_name = fw_name; + sst->dsp_ops = dsp_ops; + sst->fw_ops = bxt_fw_ops; + sst->addr.lpe = mmio_base; + sst->addr.shim = mmio_base; + + sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), + SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + + ret = skl_ipc_init(dev, skl); + if (ret) + return ret; + + skl->boot_complete = false; + init_waitqueue_head(&skl->boot_wait); + + ret = sst->fw_ops.load_fw(sst); + if (ret < 0) { + dev_err(dev, "Load base fw failed: %x", ret); + return ret; + } + + if (dsp) + *dsp = skl; + + return 0; +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); + + +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) +{ + skl_ipc_free(&ctx->ipc); + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + + if (ctx->dsp->addr.lpe) + iounmap(ctx->dsp->addr.lpe); + + ctx->dsp->ops->free(ctx->dsp); +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Broxton IPC driver"); diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 79c5089..e3d149c 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask); } +static int skl_dsp_setup_spib(struct device *dev, unsigned int size, + int stream_tag, int enable) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_stream *stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + struct hdac_ext_stream *estream; + + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + /* enable/disable SPIB for this hdac stream */ + snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + + /* set the spib value */ + snd_hdac_ext_stream_set_spib(ebus, estream, size); + + return 0; +} + +static int skl_dsp_prepare(struct device *dev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_stream *estream; + struct hdac_stream *stream; + struct snd_pcm_substream substream; + int ret; + + if (!bus) + return -ENODEV; + + memset(&substream, 0, sizeof(substream)); + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + + estream = snd_hdac_ext_stream_assign(ebus, &substream, + HDAC_EXT_STREAM_TYPE_HOST); + if (!estream) + return -ENODEV; + + stream = hdac_stream(estream); + + /* assign decouple host dma channel */ + ret = snd_hdac_dsp_prepare(stream, format, size, dmab); + if (ret < 0) + return ret; + + skl_dsp_setup_spib(dev, size, stream->stream_tag, true); + + return stream->stream_tag; +} + +static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + snd_hdac_dsp_trigger(stream, start); + + return 0; +} + +static int skl_dsp_cleanup(struct device *dev, + struct snd_dma_buffer *dmab, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_ext_stream *estream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + skl_dsp_setup_spib(dev, 0, stream_tag, false); + snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST); + + snd_hdac_dsp_cleanup(stream, dmab); + + return 0; +} + static struct skl_dsp_loader_ops skl_get_loader_ops(void) { struct skl_dsp_loader_ops loader_ops; @@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void) return loader_ops; }; +static struct skl_dsp_loader_ops bxt_get_loader_ops(void) +{ + struct skl_dsp_loader_ops loader_ops; + + memset(&loader_ops, 0, sizeof(loader_ops)); + + loader_ops.alloc_dma_buf = skl_alloc_dma_buf; + loader_ops.free_dma_buf = skl_free_dma_buf; + loader_ops.prepare = skl_dsp_prepare; + loader_ops.trigger = skl_dsp_trigger; + loader_ops.cleanup = skl_dsp_cleanup; + + return loader_ops; +}; + static const struct skl_dsp_ops dsp_ops[] = { { .id = 0x9d70, @@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = { .init = skl_sst_dsp_init, .cleanup = skl_sst_dsp_cleanup }, + { + .id = 0x5a98, + .loader_ops = bxt_get_loader_ops, + .init = bxt_sst_dsp_init, + .cleanup = bxt_sst_dsp_cleanup + }, }; static int skl_get_dsp_ops(int pci_id) diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index b6e310d..ff31e66 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -124,10 +124,19 @@ struct skl_dsp_fw_ops { }; struct skl_dsp_loader_ops { + int stream_tag; + int (*alloc_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab, size_t size); int (*free_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab); + int (*prepare)(struct device *dev, unsigned int format, + unsigned int byte_size, + struct snd_dma_buffer *bufp); + int (*trigger)(struct device *dev, bool start, int stream_tag); + + int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab, + int stream_tag); }; struct skl_load_module_info { @@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx); int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp); +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp); void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); #endif /*__SKL_SST_DSP_H__*/ -- cgit v0.10.2 From 44c376b9596ca00d1bdee37e716d1bd4dd36c955 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:02 +0900 Subject: ALSA: firewire-lib: suppress kernel warnings when releasing uninitialized stream data When any of AMDTP stream data are not initialized and private data is going to be released, WARN_ON() in amdtp_stream_destroy() is hit and dump messages. This may take users irritated. This commit fixes the bug to skip releasing when it's not initialized. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed29026..4484242 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -102,6 +102,10 @@ EXPORT_SYMBOL(amdtp_stream_init); */ void amdtp_stream_destroy(struct amdtp_stream *s) { + /* Not initialized. */ + if (s->protocol == NULL) + return; + WARN_ON(amdtp_stream_running(s)); kfree(s->protocol); mutex_destroy(&s->mutex); -- cgit v0.10.2 From 0eced45ca60666aff4c12f31fef4005ae37e859e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:03 +0900 Subject: ALSA: dice: simplify unit probe processing In former commit, ALSA dice driver doesn't generate kernel warnings when unplugging units before initializing stream data. This commit moves the initialization to delayed registration of sound card, to simplify unit probe processing. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 8b64aef..53ca441 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -201,6 +201,10 @@ static void do_registration(struct work_struct *work) dice_card_strings(dice); + err = snd_dice_stream_init_duplex(dice); + if (err < 0) + goto error; + snd_dice_create_proc(dice); err = snd_dice_create_pcm(dice); @@ -229,6 +233,7 @@ static void do_registration(struct work_struct *work) return; error: + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); snd_card_free(dice->card); dev_info(&dice->unit->device, @@ -273,12 +278,6 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) init_completion(&dice->clock_accepted); init_waitqueue_head(&dice->hwdep_wait); - err = snd_dice_stream_init_duplex(dice); - if (err < 0) { - dice_free(dice); - return err; - } - /* Allocate and register this sound card later. */ INIT_DEFERRABLE_WORK(&dice->dwork, do_registration); schedule_registration(dice); -- cgit v0.10.2 From 923f92ebb43e7a09915a5708d4805c1e099db46c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:04 +0900 Subject: ALSA: firewire-lib: add new function to schedule a work for sound card registration In former commit, ALSA dice driver postpone sound card registration after IEEE 1394 bus is calm. This idea has advantages for the other drivers. This commit adds a helper function for it to firewire-lib module. The function is really for the specific purpose. Callers should initialize delayed work structure with callback function. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 53ca441..96fe68f4 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -20,8 +20,6 @@ MODULE_LICENSE("GPL v2"); #define WEISS_CATEGORY_ID 0x00 #define LOUD_CATEGORY_ID 0x10 -#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) - /* * Some models support several isochronous channels, while these streams are not * always available. In this case, add the model name to this list. @@ -235,27 +233,12 @@ static void do_registration(struct work_struct *work) error: snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); + snd_dice_stream_destroy_duplex(dice); snd_card_free(dice->card); dev_info(&dice->unit->device, "Sound card registration failed: %d\n", err); } -static void schedule_registration(struct snd_dice *dice) -{ - struct fw_card *fw_card = fw_parent_device(dice->unit)->card; - u64 now, delay; - - now = get_jiffies_64(); - delay = fw_card->reset_jiffies + msecs_to_jiffies(PROBE_DELAY_MS); - - if (time_after64(delay, now)) - delay -= now; - else - delay = 0; - - mod_delayed_work(system_wq, &dice->dwork, delay); -} - static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_dice *dice; @@ -280,7 +263,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) /* Allocate and register this sound card later. */ INIT_DEFERRABLE_WORK(&dice->dwork, do_registration); - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); return 0; } @@ -311,7 +294,7 @@ static void dice_bus_reset(struct fw_unit *unit) /* Postpone a workqueue for deferred registration. */ if (!dice->registered) - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); /* The handler address register becomes initialized. */ snd_dice_transaction_reinit(dice); diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index f80aafa..ca4dfcf 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -67,6 +67,38 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, } EXPORT_SYMBOL(snd_fw_transaction); +#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) + +/** + * snd_fw_schedule_registration - schedule work for sound card registration + * @unit: an instance for unit on IEEE 1394 bus + * @dwork: delayed work with callback function + * + * This function is not designed for general purposes. When new unit is + * connected to IEEE 1394 bus, the bus is under bus-reset state because of + * topological change. In this state, units tend to fail both of asynchronous + * and isochronous communication. To avoid this problem, this function is used + * to postpone sound card registration after the state. The callers must + * set up instance of delayed work in advance. + */ +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork) +{ + u64 now, delay; + + now = get_jiffies_64(); + delay = fw_parent_device(unit)->card->reset_jiffies + + msecs_to_jiffies(PROBE_DELAY_MS); + + if (time_after64(delay, now)) + delay -= now; + else + delay = 0; + + mod_delayed_work(system_wq, dwork, delay); +} +EXPORT_SYMBOL(snd_fw_schedule_registration); + static void async_midi_port_callback(struct fw_card *card, int rcode, void *data, size_t length, void *callback_data) diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index f3f6f84..f676931 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -22,6 +22,9 @@ static inline bool rcode_is_permanent_error(int rcode) return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR; } +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork); + struct snd_fw_async_midi_port; typedef int (*snd_fw_async_midi_port_fill)( struct snd_rawmidi_substream *substream, -- cgit v0.10.2 From 04a2c73c97ebb224dfb411ab35bb18d7b8245e39 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:05 +0900 Subject: ALSA: bebob: delayed registration of sound card Some bebob based units tends to fail asynchronous communication when IEEE 1394 bus is under bus-reset state. When registering sound card instance at unit probe callback, userspace applications can be involved to the state. This commit postpones the registration till the bus is calm. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 932901d..f7e2cbd 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -126,6 +126,17 @@ end: return err; } +static void bebob_free(struct snd_bebob *bebob) +{ + snd_bebob_stream_destroy_duplex(bebob); + fw_unit_put(bebob->unit); + + kfree(bebob->maudio_special_quirk); + + mutex_destroy(&bebob->mutex); + kfree(bebob); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -137,18 +148,11 @@ bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; - snd_bebob_stream_destroy_duplex(bebob); - fw_unit_put(bebob->unit); - - kfree(bebob->maudio_special_quirk); - - if (bebob->card_index >= 0) { - mutex_lock(&devices_mutex); - clear_bit(bebob->card_index, devices_used); - mutex_unlock(&devices_mutex); - } + mutex_lock(&devices_mutex); + clear_bit(bebob->card_index, devices_used); + mutex_unlock(&devices_mutex); - mutex_destroy(&bebob->mutex); + bebob_free(card->private_data); } static const struct snd_bebob_spec * @@ -176,16 +180,17 @@ check_audiophile_booted(struct fw_unit *unit) return strncmp(name, "FW Audiophile Bootloader", 15) != 0; } -static int -bebob_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_bebob *bebob; - const struct snd_bebob_spec *spec; + struct snd_bebob *bebob = + container_of(work, struct snd_bebob, dwork.work); unsigned int card_index; int err; + if (bebob->registered) + return; + mutex_lock(&devices_mutex); for (card_index = 0; card_index < SNDRV_CARDS; card_index++) { @@ -193,64 +198,39 @@ bebob_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - if ((entry->vendor_id == VEN_FOCUSRITE) && - (entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH)) - spec = get_saffire_spec(unit); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH) && - !check_audiophile_booted(unit)) - spec = NULL; - else - spec = (const struct snd_bebob_spec *)entry->driver_data; - - if (spec == NULL) { - if ((entry->vendor_id == VEN_MAUDIO1) || - (entry->vendor_id == VEN_MAUDIO2)) - err = snd_bebob_maudio_load_firmware(unit); - else - err = -ENOSYS; - goto end; + err = snd_card_new(&bebob->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &bebob->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_bebob), &card); - if (err < 0) - goto end; - bebob = card->private_data; - bebob->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = bebob_card_free; - - bebob->card = card; - bebob->unit = fw_unit_get(unit); - bebob->spec = spec; - mutex_init(&bebob->mutex); - spin_lock_init(&bebob->lock); - init_waitqueue_head(&bebob->hwdep_wait); - err = name_device(bebob); if (err < 0) goto error; - if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_FW1814)) - err = snd_bebob_maudio_special_discover(bebob, true); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_PROJECTMIX)) - err = snd_bebob_maudio_special_discover(bebob, false); - else + if (bebob->spec == &maudio_special_spec) { + if (bebob->entry->model_id == MODEL_MAUDIO_FW1814) + err = snd_bebob_maudio_special_discover(bebob, true); + else + err = snd_bebob_maudio_special_discover(bebob, false); + } else { err = snd_bebob_stream_discover(bebob); + } + if (err < 0) + goto error; + + err = snd_bebob_stream_init_duplex(bebob); if (err < 0) goto error; snd_bebob_proc_init(bebob); - if ((bebob->midi_input_ports > 0) || - (bebob->midi_output_ports > 0)) { + if (bebob->midi_input_ports > 0 || bebob->midi_output_ports > 0) { err = snd_bebob_create_midi_devices(bebob); if (err < 0) goto error; @@ -264,16 +244,75 @@ bebob_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_bebob_stream_init_duplex(bebob); + err = snd_card_register(bebob->card); if (err < 0) goto error; - if (!bebob->maudio_special_quirk) { - err = snd_card_register(card); - if (err < 0) { - snd_bebob_stream_destroy_duplex(bebob); - goto error; - } + set_bit(card_index, devices_used); + mutex_unlock(&devices_mutex); + + /* + * After registered, bebob instance can be released corresponding to + * releasing the sound card instance. + */ + bebob->card->private_free = bebob_card_free; + bebob->card->private_data = bebob; + bebob->registered = true; + + return; +error: + mutex_unlock(&devices_mutex); + snd_bebob_stream_destroy_duplex(bebob); + snd_card_free(bebob->card); + dev_info(&bebob->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_bebob *bebob; + const struct snd_bebob_spec *spec; + + if (entry->vendor_id == VEN_FOCUSRITE && + entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH) + spec = get_saffire_spec(unit); + else if (entry->vendor_id == VEN_MAUDIO1 && + entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH && + !check_audiophile_booted(unit)) + spec = NULL; + else + spec = (const struct snd_bebob_spec *)entry->driver_data; + + if (spec == NULL) { + if (entry->vendor_id == VEN_MAUDIO1 || + entry->vendor_id == VEN_MAUDIO2) + return snd_bebob_maudio_load_firmware(unit); + else + return -ENODEV; + } + + /* Allocate this independent of sound card instance. */ + bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL); + if (bebob == NULL) + return -ENOMEM; + + bebob->unit = fw_unit_get(unit); + bebob->entry = entry; + bebob->spec = spec; + dev_set_drvdata(&unit->device, bebob); + + mutex_init(&bebob->mutex); + spin_lock_init(&bebob->lock); + init_waitqueue_head(&bebob->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&bebob->dwork, do_registration); + + if (entry->vendor_id != VEN_MAUDIO1 || + (entry->model_id != MODEL_MAUDIO_FW1814 && + entry->model_id != MODEL_MAUDIO_PROJECTMIX)) { + snd_fw_schedule_registration(unit, &bebob->dwork); } else { /* * This is a workaround. This bus reset seems to have an effect @@ -285,19 +324,11 @@ bebob_probe(struct fw_unit *unit, * signals from dbus and starts I/Os. To avoid I/Os till the * future bus reset, registration is done in next update(). */ - bebob->deferred_registration = true; fw_schedule_bus_reset(fw_parent_device(bebob->unit)->card, false, true); } - dev_set_drvdata(&unit->device, bebob); -end: - mutex_unlock(&devices_mutex); - return err; -error: - mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + return 0; } /* @@ -324,15 +355,11 @@ bebob_update(struct fw_unit *unit) if (bebob == NULL) return; - fcp_bus_reset(bebob->unit); - - if (bebob->deferred_registration) { - if (snd_card_register(bebob->card) < 0) { - snd_bebob_stream_destroy_duplex(bebob); - snd_card_free(bebob->card); - } - bebob->deferred_registration = false; - } + /* Postpone a workqueue for deferred registration. */ + if (!bebob->registered) + snd_fw_schedule_registration(unit, &bebob->dwork); + else + fcp_bus_reset(bebob->unit); } static void bebob_remove(struct fw_unit *unit) @@ -342,8 +369,20 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(bebob->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&bebob->dwork); + + if (bebob->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(bebob->card); + } else { + /* Don't forget this case. */ + bebob_free(bebob); + } } static const struct snd_bebob_rate_spec normal_rate_spec = { diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index b50bb33d..2a442a7 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -83,6 +83,10 @@ struct snd_bebob { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + + const struct ieee1394_device_id *entry; const struct snd_bebob_spec *spec; unsigned int midi_input_ports; @@ -111,7 +115,6 @@ struct snd_bebob { /* for M-Audio special devices */ void *maudio_special_quirk; - bool deferred_registration; /* For BeBoB version quirk. */ unsigned int version; -- cgit v0.10.2 From 7d3c1d5901aac873d13c0a03b29ee2bda183383f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:06 +0900 Subject: ALSA: fireworks: delayed registration of sound card When some fireworks units are connected sequentially, userspace applications are involved at bus-reset state on IEEE 1394 bus. In the state, any communications can be canceled. Therefore, sound card registration should be delayed till the bus gets calm. This commit achieves it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8380fb5..71a0613 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -184,6 +184,18 @@ end: return err; } +static void efw_free(struct snd_efw *efw) +{ + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); + fw_unit_put(efw->unit); + + kfree(efw->resp_buf); + + mutex_destroy(&efw->mutex); + kfree(efw); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -195,28 +207,24 @@ efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - fw_unit_put(efw->unit); - - kfree(efw->resp_buf); - if (efw->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(efw->card_index, devices_used); mutex_unlock(&devices_mutex); } - mutex_destroy(&efw->mutex); + efw_free(card->private_data); } -static int -efw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_efw *efw; - int card_index, err; + struct snd_efw *efw = container_of(work, struct snd_efw, dwork.work); + unsigned int card_index; + int err; + + if (efw->registered) + return; mutex_lock(&devices_mutex); @@ -226,24 +234,16 @@ efw_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_efw), &card); - if (err < 0) - goto end; - efw = card->private_data; - efw->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = efw_card_free; - - efw->card = card; - efw->unit = fw_unit_get(unit); - mutex_init(&efw->mutex); - spin_lock_init(&efw->lock); - init_waitqueue_head(&efw->hwdep_wait); + err = snd_card_new(&efw->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &efw->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; + } /* prepare response buffer */ snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size, @@ -260,6 +260,10 @@ efw_probe(struct fw_unit *unit, if (err < 0) goto error; + err = snd_efw_stream_init_duplex(efw); + if (err < 0) + goto error; + snd_efw_proc_init(efw); if (efw->midi_out_ports || efw->midi_in_ports) { @@ -276,44 +280,93 @@ efw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_efw_stream_init_duplex(efw); + err = snd_card_register(efw->card); if (err < 0) goto error; - err = snd_card_register(card); - if (err < 0) { - snd_efw_stream_destroy_duplex(efw); - goto error; - } - - dev_set_drvdata(&unit->device, efw); -end: + set_bit(card_index, devices_used); mutex_unlock(&devices_mutex); - return err; + + /* + * After registered, efw instance can be released corresponding to + * releasing the sound card instance. + */ + efw->card->private_free = efw_card_free; + efw->card->private_data = efw; + efw->registered = true; + + return; error: - snd_efw_transaction_remove_instance(efw); mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + snd_efw_transaction_remove_instance(efw); + snd_efw_stream_destroy_duplex(efw); + snd_card_free(efw->card); + dev_info(&efw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_efw *efw; + + efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL); + if (efw == NULL) + return -ENOMEM; + + efw->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, efw); + + mutex_init(&efw->mutex); + spin_lock_init(&efw->lock); + init_waitqueue_head(&efw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&efw->dwork, do_registration); + snd_fw_schedule_registration(unit, &efw->dwork); + + return 0; } static void efw_update(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!efw->registered) + snd_fw_schedule_registration(unit, &efw->dwork); + snd_efw_transaction_bus_reset(efw->unit); - mutex_lock(&efw->mutex); - snd_efw_stream_update_duplex(efw); - mutex_unlock(&efw->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (efw->registered) { + mutex_lock(&efw->mutex); + snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); + } } static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(efw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&efw->dwork); + + if (efw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(efw->card); + } else { + /* Don't forget this case. */ + efw_free(efw); + } } static const struct ieee1394_device_id efw_id_table[] = { diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 96c4e0c..471c772 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -65,6 +65,9 @@ struct snd_efw { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + /* for transaction */ u32 seqnum; bool resp_addr_changable; -- cgit v0.10.2 From 6c29230e2a5ff84df2b1358681414bad3e4bd220 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:07 +0900 Subject: ALSA: oxfw: delayed registration of sound card Some oxfw based units tends to fail asynchronous communication when IEEE 1394 bus is under bus-reset state. When registering sound card instance at unit probe callback, userspace applications can be involved to the state. This commit postpones the registration till the bus is calm. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index abedc22..e629b88 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -118,15 +118,8 @@ end: return err; } -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void oxfw_card_free(struct snd_card *card) +static void oxfw_free(struct snd_oxfw *oxfw) { - struct snd_oxfw *oxfw = card->private_data; unsigned int i; snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); @@ -144,6 +137,17 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ +static void oxfw_card_free(struct snd_card *card) +{ + oxfw_free(card->private_data); +} + static int detect_quirks(struct snd_oxfw *oxfw) { struct fw_device *fw_dev = fw_parent_device(oxfw->unit); @@ -205,41 +209,39 @@ static int detect_quirks(struct snd_oxfw *oxfw) return 0; } -static int oxfw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_oxfw *oxfw; + struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); int err; - if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) - return -ENODEV; + if (oxfw->registered) + return; - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*oxfw), &card); + err = snd_card_new(&oxfw->unit->device, -1, NULL, THIS_MODULE, 0, + &oxfw->card); if (err < 0) - return err; + return; - card->private_free = oxfw_card_free; - oxfw = card->private_data; - oxfw->card = card; - mutex_init(&oxfw->mutex); - oxfw->unit = fw_unit_get(unit); - oxfw->entry = entry; - spin_lock_init(&oxfw->lock); - init_waitqueue_head(&oxfw->hwdep_wait); + err = name_card(oxfw); + if (err < 0) + goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; - err = name_card(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; + if (oxfw->has_output) { + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); + if (err < 0) + goto error; + } err = snd_oxfw_create_pcm(oxfw); if (err < 0) @@ -255,54 +257,97 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); + err = snd_card_register(oxfw->card); if (err < 0) goto error; - if (oxfw->has_output) { - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); - if (err < 0) - goto error; - } - err = snd_card_register(card); - if (err < 0) { - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - goto error; - } + /* + * After registered, oxfw instance can be released corresponding to + * releasing the sound card instance. + */ + oxfw->card->private_free = oxfw_card_free; + oxfw->card->private_data = oxfw; + oxfw->registered = true; + + return; +error: + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + snd_card_free(oxfw->card); + dev_info(&oxfw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int oxfw_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_oxfw *oxfw; + + if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) + return -ENODEV; + + /* Allocate this independent of sound card instance. */ + oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL); + if (oxfw == NULL) + return -ENOMEM; + + oxfw->entry = entry; + oxfw->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, oxfw); + mutex_init(&oxfw->mutex); + spin_lock_init(&oxfw->lock); + init_waitqueue_head(&oxfw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&oxfw->dwork, do_registration); + snd_fw_schedule_registration(unit, &oxfw->dwork); + return 0; -error: - snd_card_free(card); - return err; } static void oxfw_bus_reset(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + if (!oxfw->registered) + snd_fw_schedule_registration(unit, &oxfw->dwork); + fcp_bus_reset(oxfw->unit); - mutex_lock(&oxfw->mutex); + if (oxfw->registered) { + mutex_lock(&oxfw->mutex); - snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); + snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); - mutex_unlock(&oxfw->mutex); + mutex_unlock(&oxfw->mutex); - if (oxfw->entry->vendor_id == OUI_STANTON) - snd_oxfw_scs1x_update(oxfw); + if (oxfw->entry->vendor_id == OUI_STANTON) + snd_oxfw_scs1x_update(oxfw); + } } static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(oxfw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&oxfw->dwork); + + if (oxfw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(oxfw->card); + } else { + /* Don't forget this case. */ + oxfw_free(oxfw); + } } static const struct compat_info griffin_firewave = { diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 2c84714e..2047dcb 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -39,6 +39,9 @@ struct snd_oxfw { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; -- cgit v0.10.2 From 86c8dd7f4da3fb3f92fc5ab5144c971639d39745 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:08 +0900 Subject: ALSA: firewire-digi00x: delayed registration of sound card When some digi00x units are connected sequentially, userspace applications are involved at bus-reset state on IEEE 1394 bus. In the state, any communications can be canceled. Therefore, sound card registration should be delayed till the bus gets calm. This commit achieves it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c index 554324d..735d356 100644 --- a/sound/firewire/digi00x/digi00x-transaction.c +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -126,12 +126,17 @@ int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) return err; error: fw_core_remove_address_handler(&dg00x->async_handler); - dg00x->async_handler.address_callback = NULL; + dg00x->async_handler.callback_data = NULL; return err; } void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x) { + if (dg00x->async_handler.callback_data == NULL) + return; + snd_fw_async_midi_port_destroy(&dg00x->out_control); fw_core_remove_address_handler(&dg00x->async_handler); + + dg00x->async_handler.callback_data = NULL; } diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f33b7a..cc4776c 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -40,10 +40,8 @@ static int name_card(struct snd_dg00x *dg00x) return 0; } -static void dg00x_card_free(struct snd_card *card) +static void dg00x_free(struct snd_dg00x *dg00x) { - struct snd_dg00x *dg00x = card->private_data; - snd_dg00x_stream_destroy_duplex(dg00x); snd_dg00x_transaction_unregister(dg00x); @@ -52,28 +50,24 @@ static void dg00x_card_free(struct snd_card *card) mutex_destroy(&dg00x->mutex); } -static int snd_dg00x_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void dg00x_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_dg00x *dg00x; - int err; + dg00x_free(card->private_data); +} - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_dg00x), &card); - if (err < 0) - return err; - card->private_free = dg00x_card_free; +static void do_registration(struct work_struct *work) +{ + struct snd_dg00x *dg00x = + container_of(work, struct snd_dg00x, dwork.work); + int err; - /* initialize myself */ - dg00x = card->private_data; - dg00x->card = card; - dg00x->unit = fw_unit_get(unit); + if (dg00x->registered) + return; - mutex_init(&dg00x->mutex); - spin_lock_init(&dg00x->lock); - init_waitqueue_head(&dg00x->hwdep_wait); + err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0, + &dg00x->card); + if (err < 0) + return; err = name_card(dg00x); if (err < 0) @@ -101,35 +95,86 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(dg00x->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, dg00x); + dg00x->card->private_free = dg00x_card_free; + dg00x->card->private_data = dg00x; + dg00x->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_dg00x_transaction_unregister(dg00x); + snd_dg00x_stream_destroy_duplex(dg00x); + snd_card_free(dg00x->card); + dev_info(&dg00x->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_dg00x_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_dg00x *dg00x; + + /* Allocate this independent of sound card instance. */ + dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL); + if (dg00x == NULL) + return -ENOMEM; + + dg00x->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, dg00x); + + mutex_init(&dg00x->mutex); + spin_lock_init(&dg00x->lock); + init_waitqueue_head(&dg00x->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&dg00x->dwork, do_registration); + snd_fw_schedule_registration(unit, &dg00x->dwork); + + return 0; } static void snd_dg00x_update(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!dg00x->registered) + snd_fw_schedule_registration(unit, &dg00x->dwork); + snd_dg00x_transaction_reregister(dg00x); - mutex_lock(&dg00x->mutex); - snd_dg00x_stream_update_duplex(dg00x); - mutex_unlock(&dg00x->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (dg00x->registered) { + mutex_lock(&dg00x->mutex); + snd_dg00x_stream_update_duplex(dg00x); + mutex_unlock(&dg00x->mutex); + } } static void snd_dg00x_remove(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dg00x->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&dg00x->dwork); + + if (dg00x->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(dg00x->card); + } else { + /* Don't forget this case. */ + dg00x_free(dg00x); + } } static const struct ieee1394_device_id snd_dg00x_id_table[] = { diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 907e739..2cd465c 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -37,6 +37,9 @@ struct snd_dg00x { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + struct amdtp_stream tx_stream; struct fw_iso_resources tx_resources; -- cgit v0.10.2 From 44fde3b89ba1e154b3cec7d711703fff53852983 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 4 Apr 2016 19:23:54 +0530 Subject: ALSA: hda - Update chmap tlv to report sink's capability The existing TLV callback implementation copies all of the cea_channel_speaker_allocation map table to the TLV container irrespective of what is reported by sink. This is of little use to the userspace application. With this patch, it parses the spk_alloc block as queried from the ELD, and copies only the corresponding mapping channel allocation entries from the cea channel speaker allocation table. Thus the user can parse the TLV container to identify sink's capability and set the channel map accordingly. It shouldn't impact the behavior in AMD chipset, as this makes use of already parsed spk alloc block to calculate the channel map. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h index e20d219..babd445 100644 --- a/include/sound/hda_chmap.h +++ b/include/sound/hda_chmap.h @@ -36,6 +36,8 @@ struct hdac_chmap_ops { int (*chmap_validate)(struct hdac_chmap *hchmap, int ca, int channels, unsigned char *chmap); + int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx); + void (*get_chmap)(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap); void (*set_chmap)(struct hdac_device *hdac, int pcm_idx, diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec862..c6c75e7 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static int spk_mask_from_spk_alloc(int spk_alloc) +{ + int i; + int spk_mask = eld_speaker_allocation_bits[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + return spk_mask; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; int chs, count = 0; + unsigned long max_chs; + int type; + int spk_alloc, spk_mask; if (size < 8) return -ENOMEM; @@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + spk_mask = spk_mask_from_spk_alloc(spk_alloc); + + max_chs = hweight_long(spk_mask); + + for (chs = 2; chs <= max_chs; chs++) { int i; struct hdac_cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (cap->channels != chs) + continue; + + if (!(cap->spk_mask == (spk_mask & cap->spk_mask))) continue; + + type = chmap->ops.chmap_cea_alloc_validate_get_type( + chmap, cap, chs); + if (type < 0) + return -ENODEV; if (size < 8) return -ENOMEM; + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; + dst += 2; size -= 8; count += 8; + if (size < chs_bytes) return -ENOMEM; + size -= chs_bytes; count += chs_bytes; chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) return -EFAULT; dst += chs; } } + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7b..9452384 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1837,6 +1837,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2165,6 +2177,7 @@ static int alloc_generic_hdmi(struct hda_codec *codec) spec->chmap.ops.get_chmap = hdmi_get_chmap; spec->chmap.ops.set_chmap = hdmi_set_chmap; spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, codec->spec = spec; hdmi_array_init(spec, 4); -- cgit v0.10.2 From 613c7c4003c8338a9a638485d95de2775948295b Mon Sep 17 00:00:00 2001 From: Jose Abreu Date: Tue, 5 Apr 2016 18:08:02 +0100 Subject: ASoC: dwc: Unmask I2S interrupts only for enabled channels There is no need to unmask all interrupts at I2S start. This can cause performance issues in slower platforms. Unmask only the interrupts for the used channels. Signed-off-by: Jose Abreu Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bff258d..3effcd1 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -147,17 +147,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { + struct i2s_clk_config_data *config = &dev->config; u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); } i2s_write_reg(dev->i2s_base, ITER, 1); } else { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); } -- cgit v0.10.2 From 4926c8046549cc3c9689e8050e303c016a0b0cba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Apr 2016 11:33:54 +0200 Subject: ALSA: intel8x0: Drop superfluous VM checks intel8x0 driver has the inside_vm check for skipping a buggy hardware workaround in the PCM pointer callback in the commit [228cf79376f1: ALSA: intel8x0: Improve performance in virtual environment]. This was originally applied to all devices on known VMs, but the code was switched to use the PCI ID matching for applying to only known devices (KVM and Parallels), in order to avoid applying wrongly to VT-d and other such cases, in the commit [7fb4f392bd27: ALSA: intel8x0: improve virtual environment detection]. Meanwhile, the original VM check was kept even after switching to the PCI ID matching. It was partly because we weren't 100% sure whether we had covered all well, and partly because this would help identifying the issue once when a user of another VM hit the same problem or a regression. Currently the VM check is used only for showing the kernel message that the VM-optimization isn't applied, and the VM check itself doesn't change the actual driver behavior at all. Despite the relatively safe driver behavior, the code caught attention of developers badly and brought many confusion / misunderstanding. Since we've got neither regression nor enhancement report for other VMs for five years long, it's likely safe to drop this superfluous VM check now. The module option is still kept, so if a user still needs to adjust, it can be applied as was. Acked-by: Konstantin Ozerkov Signed-off-by: Takashi Iwai diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8151318..9720a30 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -42,12 +42,6 @@ #include #include -#ifdef CONFIG_KVM_GUEST -#include -#else -#define kvm_para_available() (0) -#endif - MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); @@ -2972,25 +2966,17 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) goto fini; } - /* detect KVM and Parallels virtual environments */ - result = kvm_para_available(); -#ifdef X86_FEATURE_HYPERVISOR - result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); -#endif - if (!result) - goto fini; - /* check for known (emulated) devices */ + result = 0; if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET && pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) { /* KVM emulated sound, PCI SSID: 1af4:1100 */ msg = "enable KVM"; + result = 1; } else if (pci->subsystem_vendor == 0x1ab8) { /* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */ msg = "enable Parallels VM"; - } else { - msg = "disable (unknown or VT-d) VM"; - result = 0; + result = 1; } fini: -- cgit v0.10.2 From 9771b18a0b374b6e6ecfa84c8b59d5ef79e969b1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 6 Apr 2016 11:21:53 +0100 Subject: ASoC: wm_adsp: Factor out fetching of stream errors from the DSP Factor out the reading of the DSP error flag into its own function to support further improvements to the code. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 953c427..f70c609 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2816,6 +2816,23 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) return 0; } +static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf) +{ + int ret; + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); + if (ret < 0) { + adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret); + return ret; + } + if (buf->error != 0) { + adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error); + return -EIO; + } + + return 0; +} + int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) { struct wm_adsp_compr_buf *buf; @@ -2834,16 +2851,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) adsp_dbg(dsp, "Handling buffer IRQ\n"); - ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); - if (ret < 0) { - adsp_err(dsp, "Failed to check buffer error: %d\n", ret); - goto out; - } - if (buf->error != 0) { - adsp_err(dsp, "Buffer error occurred: %d\n", buf->error); - ret = -EIO; + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) goto out; - } ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); -- cgit v0.10.2 From 5847609edb3c80be07e897e449a9bb579a0fe9d8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 6 Apr 2016 11:21:54 +0100 Subject: ASoC: wm_adsp: Improve DSP error handling If we encounter an error on the DSP side whilst user-space is waiting on the poll we should call snd_compr_fragment_elapsed, although data is not actually available we want to wake user-space such that the error can be propagated out quickly. Additionally some versions of the DSP firmware are not super consistent about actually generating an IRQ if they encounter an error, as such we will check the DSP error status every time we run out of available data as well, to ensure we catch it. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f70c609..3ac2e1f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2853,7 +2853,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) ret = wm_adsp_buffer_get_error(buf); if (ret < 0) - goto out; + goto out_notify; /* Wake poll to report error */ ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); @@ -2868,6 +2868,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) goto out; } +out_notify: if (compr && compr->stream) snd_compr_fragment_elapsed(compr->stream); @@ -2928,6 +2929,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, * DSP to inform us once a whole fragment is available. */ if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) + goto out; + ret = wm_adsp_buffer_reenable_irq(buf); if (ret < 0) { adsp_err(dsp, -- cgit v0.10.2 From 4a4436573a6669516f73bac25016683d396ed4c4 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 31 Mar 2016 16:35:58 +0300 Subject: ALSA: pcm: add IEC958 channel status helper for hw_params Add IEC958 channel status helper that gets the audio properties from snd_pcm_hw_params instead of snd_pcm_runtime. This is needed to produce the channel status bits already in audio stream configuration phase. Signed-off-by: Jyri Sarha Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/include/sound/pcm_iec958.h b/include/sound/pcm_iec958.h index 0eed397..36f023a 100644 --- a/include/sound/pcm_iec958.h +++ b/include/sound/pcm_iec958.h @@ -6,4 +6,6 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, size_t len); +int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params, + u8 *cs, size_t len); #endif diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c index 36b2d7a..e016871 100644 --- a/sound/core/pcm_iec958.c +++ b/sound/core/pcm_iec958.c @@ -9,30 +9,18 @@ #include #include #include +#include #include -/** - * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status - * @runtime: pcm runtime structure with ->rate filled in - * @cs: channel status buffer, at least four bytes - * @len: length of channel status buffer - * - * Create the consumer format channel status data in @cs of maximum size - * @len corresponding to the parameters of the PCM runtime @runtime. - * - * Drivers may wish to tweak the contents of the buffer after creation. - * - * Returns: length of buffer, or negative error code if something failed. - */ -int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, - size_t len) +static int create_iec958_consumer(uint rate, uint sample_width, + u8 *cs, size_t len) { unsigned int fs, ws; if (len < 4) return -EINVAL; - switch (runtime->rate) { + switch (rate) { case 32000: fs = IEC958_AES3_CON_FS_32000; break; @@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, } if (len > 4) { - switch (snd_pcm_format_width(runtime->format)) { + switch (sample_width) { case 16: ws = IEC958_AES4_CON_WORDLEN_20_16; break; @@ -92,4 +80,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, return len; } + +/** + * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status + * @runtime: pcm runtime structure with ->rate filled in + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, + size_t len) +{ + return create_iec958_consumer(runtime->rate, + snd_pcm_format_width(runtime->format), + cs, len); +} EXPORT_SYMBOL(snd_pcm_create_iec958_consumer); + +/** + * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status + * @hw_params: the hw_params instance for extracting rate and sample format + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params, + u8 *cs, size_t len) +{ + return create_iec958_consumer(params_rate(params), params_width(params), + cs, len); +} +EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params); -- cgit v0.10.2 From 4a462ce084d5beb92cfc68f53f88c035c82e6b59 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 31 Mar 2016 16:35:59 +0300 Subject: ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper Treat 32 bit sample width as if it was 24 bits when generating IEC958 channel status bits. On some platforms 24 sample width is problematic and to get full 24 bit precision a 32 bit format, using only the 24 most significant bits, may have to be used. Signed-off-by: Jyri Sarha Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c index e016871..5e6aed6 100644 --- a/sound/core/pcm_iec958.c +++ b/sound/core/pcm_iec958.c @@ -59,6 +59,7 @@ static int create_iec958_consumer(uint rate, uint sample_width, IEC958_AES4_CON_MAX_WORDLEN_24; break; case 24: + case 32: /* Assume 24-bit width for 32-bit samples. */ ws = IEC958_AES4_CON_WORDLEN_24_20 | IEC958_AES4_CON_MAX_WORDLEN_24; break; -- cgit v0.10.2 From 3fafd14d9422c46f5c2a142298384dc15dbf88b2 Mon Sep 17 00:00:00 2001 From: Jose Abreu Date: Thu, 7 Apr 2016 17:53:57 +0100 Subject: ASoC: dwc: Use fifo depth to program FCR This patch makes Designware I2S driver use the fifo depth value to program the fifo configuration register instead of using hardcoded values. Signed-off-by: Jose Abreu Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 3effcd1..0db69b7 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -100,6 +100,7 @@ struct dw_i2s_dev { struct device *dev; u32 ccr; u32 xfer_resolution; + u32 fifo_th; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; @@ -232,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); } else { i2s_write_reg(dev->i2s_base, RCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); @@ -499,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, */ u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1); u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2); + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); u32 idx; if (dev->capability & DWC_I2S_RECORD && @@ -537,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, dev->capability |= DW_I2S_SLAVE; } + dev->fifo_th = fifo_depth / 2; return 0; } -- cgit v0.10.2 From e8369d65693bd4e21e043c0e66eff1056ed1e7a3 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 8 Apr 2016 12:45:25 +0900 Subject: ALSA: Fix a typo in timestamping.txt This patch fix a spelling typo found in Documentation/sound/alsa/timestamping.txt Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt index 0b191a2..1b6473f 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an interpolation of the results In some hardware-specific configuration, the system timestamp is -latched by a low-level audio subsytem, and the information provided +latched by a low-level audio subsystem, and the information provided back to the driver. Due to potential delays in the communication with the hardware, there is a risk of misalignment with the avail and delay information. To make sure applications are not confused, a -- cgit v0.10.2 From 5305239312a5fcc50849e157a3178778c6914aa0 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 9 Apr 2016 10:36:15 +0200 Subject: ALSA: constify ct_timer_ops structures The ct_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d4604..8f94534 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop, -- cgit v0.10.2 From cddaafb9a4a7b6d19030d2632107ba2aa068474d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 9 Apr 2016 11:21:46 +0200 Subject: ALSA: usb-audio: add UAC2 clock sources as mixer controls UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016). Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..cab6f52 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include #include #include +#include #include #include #include @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- cgit v0.10.2 From 191227d99a281333b50aaf82ab4152fbfa249c19 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 8 Apr 2016 19:52:02 +0200 Subject: ALSA: usb-audio: allow clock source validity interrupts miniDSP USBStreamer UAC2 devices send clock validity changes with the control field set to zero. The current interrupt handler ignores all packets if the control field does not match the mixer element's, but it really should only do that in case that field is needed to distinguish multiple elements with the same ID. This patch implements a logic that lets notifications packets pass if the element ID is unique for a given device. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index cab6f52..2f8c388 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2374,6 +2374,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, __u8 unitid = (index >> 8) & 0xff; __u8 control = (value >> 8) & 0xff; __u8 channel = value & 0xff; + unsigned int count = 0; if (channel >= MAX_CHANNELS) { usb_audio_dbg(mixer->chip, @@ -2382,6 +2383,12 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + count++; + + if (count == 0) + return; + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { struct usb_mixer_elem_info *info; @@ -2389,7 +2396,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, continue; info = (struct usb_mixer_elem_info *)list; - if (info->control != control) + if (count > 1 && info->control != control) continue; switch (attribute) { -- cgit v0.10.2 From 43b27d7286737d9af9ebeff0219e38560cb31748 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 8 Apr 2016 16:50:14 +0100 Subject: ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831 The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 92d22a0..d8a6823 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- cgit v0.10.2 From 4f29efc0ea61de1482a27c580575d860385cd54f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Apr 2016 15:16:24 +0200 Subject: ALSA: hda - Add missing capture_hook calls for dyn-ADC PCM streams The calls for capture_hook were missing in dyn_adc_capture_pcm_prepare and cleanup callbacks. Luckily there are no users of the capture hooks with dyn-adc PCM, so far, thus this doesn't change the behavior of existing devices, but it's a fix for a future usage. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89..dc2c136 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5432,6 +5432,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE); return 0; } @@ -5442,6 +5443,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_gen_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP); return 0; } -- cgit v0.10.2 From fa44b7ec9bc4115513e59f31da1167166bd6346a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Apr 2016 15:23:05 +0200 Subject: ALSA: hda - Update documentation Update the URLs for alsa-info.sh and hda-emu. Also drop the alsa-driver snapshot URL since it's been discontinued recently. Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index e7193aa..d4510eb 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -655,17 +655,6 @@ development branches in general while the development for the current and next kernels are found in for-linus and for-next branches, respectively. -If you are using the latest Linus tree, it'd be better to pull the -above GIT tree onto it. If you are using the older kernels, an easy -way to try the latest ALSA code is to build from the snapshot -tarball. There are daily tarballs and the latest snapshot tarball. -All can be built just like normal alsa-driver release packages, that -is, installed via the usual spells: configure, make and make -install(-modules). See INSTALL in the package. The snapshot tarballs -are found at: - -- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ - Sending a Bug Report ~~~~~~~~~~~~~~~~~~~~ @@ -699,7 +688,12 @@ problems. alsa-info ~~~~~~~~~ The script `alsa-info.sh` is a very useful tool to gather the audio -device information. You can fetch the latest version from: +device information. It's included in alsa-utils package. The latest +version can be found on git repository: + +- git://git.alsa-project.org/alsa-utils.git + +The script can be fetched directly from the following URL, too: - http://www.alsa-project.org/alsa-info.sh @@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control (mixer) elements, set/get the control element value, simulate the PCM operation, the jack plugging simulation, etc. -The package is found in: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -A git repository is available: +The program is found in the git repository below: - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git -See README file in the tarball for more details about hda-emu +See README file in the repository for more details about hda-emu program. -- cgit v0.10.2 From e92077c3f45395881ad8c690bb86a85ffe5198ba Mon Sep 17 00:00:00 2001 From: Heinrich Schuchardt Date: Tue, 12 Apr 2016 22:51:03 +0200 Subject: ASoC: fsl: imx-pcm-fiq: use correct format specifier Documentation/printk-formats.txt has size_t: use %zu or %zx runtime->dma_bytes is of type size_t. Signed-off-by: Heinrich Schuchardt Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index e63cd5e..dac6688 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret, runtime->dma_area, &runtime->dma_addr, runtime->dma_bytes); -- cgit v0.10.2 From 896491b304f956d3e87208a242db4fdfa952cdc5 Mon Sep 17 00:00:00 2001 From: Heinrich Schuchardt Date: Wed, 13 Apr 2016 01:54:01 +0200 Subject: ASoC: au1x: use correct format specifier Documentation/printk-formats.txt has unsigned long: use %lu or %lx size_t: use %zu or %zx runtime->dma_bytes is of type size_t. runtime->min_align is of type unsigned long. Signed-off-by: Heinrich Schuchardt Signed-off-by: Mark Brown diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 5741c0a..b5d1caa 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, stype = substream->stream; pcd = to_dmadata(substream); - DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " - "runtime->min_align %d\n", + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", (unsigned long)runtime->dma_area, (unsigned long)runtime->dma_addr, runtime->dma_bytes, runtime->min_align); -- cgit v0.10.2 From 3aa02cb664c5fb1042958c8d1aa8c35055a2ebc4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Apr 2016 18:02:37 +0200 Subject: ALSA: pcm : Call kill_fasync() in stream lock Currently kill_fasync() is called outside the stream lock in snd_pcm_period_elapsed(). This is potentially racy, since the stream may get released even during the irq handler is running. Although snd_pcm_release_substream() calls snd_pcm_drop(), this doesn't guarantee that the irq handler finishes, thus the kill_fasync() call outside the stream spin lock may be invoked after the substream is detached, as recently reported by KASAN. As a quick workaround, move kill_fasync() call inside the stream lock. The fasync is rarely used interface, so this shouldn't have a big impact from the performance POV. Ideally, we should implement some sync mechanism for the proper finish of stream and irq handler. But this oneliner should suffice for most cases, so far. Reported-by: Baozeng Ding Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3a9b66c..0aca397 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_timer_interrupt(substream->timer, 1); #endif _end: - snd_pcm_stream_unlock_irqrestore(substream, flags); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); -- cgit v0.10.2 From a19c921fca0a865b657d59b2c9a05aa0a2905126 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Apr 2016 15:28:52 +0200 Subject: ALSA: lx646es: Fix possible uninitialized variable reference lx_pipe_state() checks the return value from lx_message_send_atomic() and breaks the loop only when it's a negative value. However, lx_message_send_atomic() may return a positive error code (as the return code from the hardware), and then lx_pipe_state() tries to compare the uninitialized current_state variable. Fix this behavior by checking the positive non-zero error code as well. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index f3d6202..a80684b 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -644,7 +644,7 @@ static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, if (err < 0) return err; - if (current_state == state) + if (!err && current_state == state) return 0; mdelay(1); -- cgit v0.10.2 From 22225835e23f7a768e767967004d2f3751c64be5 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Mon, 18 Apr 2016 14:32:40 +0200 Subject: ASoC: davinci-mcbsp: add binding for McBSP Add devicetree binding for the TI DA850/OMAP-L138/AM18xx MultiChannel Buffered Serial Port (McBSP) The optional register range "dat" is not implemented at the moment. The current driver supports only DMA into RX/TX registers but no FIFO. Once the FIFO is implemented in the driver the "dat" range will be used. Signed-off-by: Petr Kulhavy Reviewed-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt new file mode 100644 index 0000000..55b53e1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt @@ -0,0 +1,51 @@ +Texas Instruments DaVinci McBSP module +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This binding describes the "Multi-channel Buffered Serial Port" (McBSP) +audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x. + + +Required properties: +~~~~~~~~~~~~~~~~~~~~ +- compatible : + "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms + +- reg : physical base address and length of the controller memory mapped + region(s). +- reg-names : Should contain: + * "mpu" for the main registers (required). + * "dat" for the data FIFO (optional). + +- dmas: three element list of DMA controller phandles, DMA request line and + TC channel ordered triplets. +- dma-names: identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. The dma + identifiers must be "rx" and "tx". + +Optional properties: +~~~~~~~~~~~~~~~~~~~~ +- interrupts : Interrupt numbers for McBSP +- interrupt-names : Known interrupt names are "rx" and "tx" + +- pinctrl-0: Should specify pin control group used for this controller. +- pinctrl-names: Should contain only one value - "default", for more details + please refer to pinctrl-bindings.txt + +Example (AM1808): +~~~~~~~~~~~~~~~~~ + +mcbsp0: mcbsp@1d10000 { + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x00110000 0x1000>, + <0x00310000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97 98>; + interrupts-names = "rx", "tx"; + dmas = <&edma0 3 1 + &edma0 2 1>; + dma-names = "tx", "rx"; + status = "okay"; +}; -- cgit v0.10.2 From 5f9a50c3e55ee887b7a0ccb68045b92579972b55 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Mon, 18 Apr 2016 14:32:41 +0200 Subject: ASoC: Davinci: McBSP: add device tree support for McBSP This adds DT support for the TI DA8xx/OMAP-L1x/AM17xx/AM18xx McBSP driver. Signed-off-by: Petr Kulhavy Reviewed-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50ca291..6b732d8 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -16,7 +16,11 @@ config SND_EDMA_SOC - DRA7xx family config SND_DAVINCI_SOC_I2S - tristate + tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support" + depends on SND_EDMA_SOC + help + Say Y or M here if you want to have support for McBSP IP found in + Texas Instruments DaVinci DA850 SoCs. config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ec98548..3849616 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -4,9 +4,15 @@ * Author: Vladimir Barinov, * Copyright: (C) 2007 MontaVista Software, Inc., * + * DT support (c) 2016 Petr Kulhavy, Barix AG + * based on davinci-mcasp.c DT support + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. + * + * TODO: + * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers */ #include @@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { + struct snd_dmaengine_dai_dma_data *dma_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *res; void __iomem *io_base; int *dma; int ret; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!mem) { + dev_warn(&pdev->dev, + "\"mpu\" mem resource not found, using index 0\n"); + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + } + io_base = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(io_base)) return PTR_ERR(io_base); @@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return -ENODEV; - clk_enable(dev->clk); - dev->base = io_base; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); + /* setup DMA, first TX, then RX */ + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); - - /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "tx"; + } else { + dev_err(&pdev->dev, "Missing DMA tx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; + + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "rx"; + } else { + dev_err(&pdev->dev, "Missing DMA rx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return -ENODEV; + clk_enable(dev->clk); dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id davinci_i2s_match[] = { + { .compatible = "ti,da850-mcbsp" }, + {}, +}; +MODULE_DEVICE_TABLE(of, davinci_i2s_match); + static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", + .of_match_table = of_match_ptr(davinci_i2s_match), }, }; -- cgit v0.10.2 From 09184118a8abae030539469848d475adcc0e5839 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 31 Mar 2016 16:36:00 +0300 Subject: ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders The hdmi-codec is a platform device driver to be registered from drivers of external HDMI encoders with I2S and/or spdif interface. The driver in turn registers an ASoC codec for the HDMI encoder's audio functionality. The structures and definitions in the API header are mostly redundant copies of similar structures in ASoC headers. This is on purpose to avoid direct dependencies to ASoC structures in video side driver. Signed-off-by: Jyri Sarha Acked-by: Arnaud Pouliquen Acked-by: PC Liao Signed-off-by: Mark Brown diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h new file mode 100644 index 0000000..fc3a481 --- /dev/null +++ b/include/sound/hdmi-codec.h @@ -0,0 +1,100 @@ +/* + * hdmi-codec.h - HDMI Codec driver API + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Jyri Sarha + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __HDMI_CODEC_H__ +#define __HDMI_CODEC_H__ + +#include +#include +#include +#include + +/* + * Protocol between ASoC cpu-dai and HDMI-encoder + */ +struct hdmi_codec_daifmt { + enum { + HDMI_I2S, + HDMI_RIGHT_J, + HDMI_LEFT_J, + HDMI_DSP_A, + HDMI_DSP_B, + HDMI_AC97, + HDMI_SPDIF, + } fmt; + int bit_clk_inv:1; + int frame_clk_inv:1; + int bit_clk_master:1; + int frame_clk_master:1; +}; + +/* + * HDMI audio parameters + */ +struct hdmi_codec_params { + struct hdmi_audio_infoframe cea; + struct snd_aes_iec958 iec; + int sample_rate; + int sample_width; + int channels; +}; + +struct hdmi_codec_ops { + /* + * Called when ASoC starts an audio stream setup. + * Optional + */ + int (*audio_startup)(struct device *dev); + + /* + * Configures HDMI-encoder for audio stream. + * Mandatory + */ + int (*hw_params)(struct device *dev, + struct hdmi_codec_daifmt *fmt, + struct hdmi_codec_params *hparms); + + /* + * Shuts down the audio stream. + * Mandatory + */ + void (*audio_shutdown)(struct device *dev); + + /* + * Mute/unmute HDMI audio stream. + * Optional + */ + int (*digital_mute)(struct device *dev, bool enable); + + /* + * Provides EDID-Like-Data from connected HDMI device. + * Optional + */ + int (*get_eld)(struct device *dev, uint8_t *buf, size_t len); +}; + +/* HDMI codec initalization data */ +struct hdmi_codec_pdata { + const struct hdmi_codec_ops *ops; + uint i2s:1; + uint spdif:1; + int max_i2s_channels; +}; + +#define HDMI_CODEC_DRV_NAME "hdmi-audio-codec" + +#endif /* __HDMI_CODEC_H__ */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..06d0e05 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -88,6 +88,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_NAU8825 if I2C + select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM179X_I2C if I2C select SND_SOC_PCM179X_SPI if SPI_MASTER @@ -477,6 +478,11 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..d7185dd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -81,6 +81,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o +snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm179x-codec-objs := pcm179x.o snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c new file mode 100644 index 0000000..b46b8ed --- /dev/null +++ b/sound/soc/codecs/hdmi-codec.c @@ -0,0 +1,396 @@ +/* + * ALSA SoC codec for HDMI encoder drivers + * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Jyri Sarha + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include /* This is only to get MAX_ELD_BYTES */ + +struct hdmi_codec_priv { + struct hdmi_codec_pdata hcd; + struct snd_soc_dai_driver *daidrv; + struct hdmi_codec_daifmt daifmt[2]; + struct mutex current_stream_lock; + struct snd_pcm_substream *current_stream; + struct snd_pcm_hw_constraint_list ratec; + uint8_t eld[MAX_ELD_BYTES]; +}; + +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +enum { + DAI_ID_I2S = 0, + DAI_ID_SPDIF, +}; + +static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + mutex_lock(&hcp->current_stream_lock); + if (!hcp->current_stream) { + hcp->current_stream = substream; + } else if (hcp->current_stream != substream) { + dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + ret = -EINVAL; + } + mutex_unlock(&hcp->current_stream_lock); + + return ret; +} + +static int hdmi_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + if (hcp->hcd.ops->audio_startup) { + ret = hcp->hcd.ops->audio_startup(dai->dev->parent); + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); + return ret; + } + } + + if (hcp->hcd.ops->get_eld) { + ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld, + sizeof(hcp->eld)); + + if (!ret) { + ret = snd_pcm_hw_constraint_eld(substream->runtime, + hcp->eld); + if (ret) + return ret; + } + } + return 0; +} + +static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + WARN_ON(hcp->current_stream != substream); + + hcp->hcd.ops->audio_shutdown(dai->dev->parent); + + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); +} + +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_params hp = { + .iec = { + .status = { 0 }, + .subcode = { 0 }, + .pad = 0, + .dig_subframe = { 0 }, + } + }; + int ret; + + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, + params_width(params), params_rate(params), + params_channels(params)); + + if (params_width(params) > 24) + params->msbits = 24; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, + sizeof(hp.iec.status)); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + hdmi_audio_infoframe_init(&hp.cea); + hp.cea.channels = params_channels(params); + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; + + hp.sample_width = params_width(params); + hp.sample_rate = params_rate(params); + hp.channels = params_channels(params); + + return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id], + &hp); +} + +static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_daifmt cf = { 0 }; + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (dai->id == DAI_ID_SPDIF) { + cf.fmt = HDMI_SPDIF; + } else { + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + } + + hcp->daifmt[dai->id] = cf; + + return ret; +} + +static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (hcp->hcd.ops->digital_mute) + return hcp->hcd.ops->digital_mute(dai->dev->parent, mute); + + return 0; +} + +static const struct snd_soc_dai_ops hdmi_dai_ops = { + .startup = hdmi_codec_startup, + .shutdown = hdmi_codec_shutdown, + .hw_params = hdmi_codec_hw_params, + .set_fmt = hdmi_codec_set_fmt, + .digital_mute = hdmi_codec_digital_mute, +}; + + +#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +/* + * This list is only for formats allowed on the I2S bus. So there is + * some formats listed that are not supported by HDMI interface. For + * instance allowing the 32-bit formats enables 24-precision with CPU + * DAIs that do not support 24-bit formats. If the extra formats cause + * problems, we should add the video side driver an option to disable + * them. + */ +#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", + .id = DAI_ID_I2S, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = HDMI_RATES, + .formats = I2S_FORMATS, + .sig_bits = 24, + }, + .ops = &hdmi_dai_ops, +}; + +static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", + .id = DAI_ID_SPDIF, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = HDMI_RATES, + .formats = SPDIF_FORMATS, + }, + .ops = &hdmi_dai_ops, +}; + +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + +static int hdmi_codec_probe(struct platform_device *pdev) +{ + struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; + struct device *dev = &pdev->dev; + struct hdmi_codec_priv *hcp; + int dai_count, i = 0; + int ret; + + dev_dbg(dev, "%s()\n", __func__); + + if (!hcd) { + dev_err(dev, "%s: No plalform data\n", __func__); + return -EINVAL; + } + + dai_count = hcd->i2s + hcd->spdif; + if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params || + !hcd->ops->audio_shutdown) { + dev_err(dev, "%s: Invalid parameters\n", __func__); + return -EINVAL; + } + + hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL); + if (!hcp) + return -ENOMEM; + + hcp->hcd = *hcd; + mutex_init(&hcp->current_stream_lock); + + hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + GFP_KERNEL); + if (!hcp->daidrv) + return -ENOMEM; + + if (hcd->i2s) { + hcp->daidrv[i] = hdmi_i2s_dai; + hcp->daidrv[i].playback.channels_max = + hcd->max_i2s_channels; + i++; + } + + if (hcd->spdif) + hcp->daidrv[i] = hdmi_spdif_dai; + + ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, + dai_count); + if (ret) { + dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n", + __func__, ret); + return ret; + } + + dev_set_drvdata(dev, hcp); + return 0; +} + +static int hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_codec_driver = { + .driver = { + .name = HDMI_CODEC_DRV_NAME, + }, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, +}; + +module_platform_driver(hdmi_codec_driver); + +MODULE_AUTHOR("Jyri Sarha "); +MODULE_DESCRIPTION("HDMI Audio Codec Driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME); -- cgit v0.10.2 From d23f0517357ef48d2845846e899d125b3c1a492e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Apr 2016 15:28:39 +0200 Subject: ALSA: ens1371: Fix "Line In->Rear Out Switch" control The "Line In->Rear Out Switch" control on ens1371 driver returns a bogus value, always true, as its check is totally broken. Fix it to check the proper GPIO bit mask. Reported-by: David Binderman Signed-off-by: Takashi Iwai diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 0dc44eb..626cd21 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1548,7 +1548,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, int val = 0; spin_lock_irq(&ensoniq->reg_lock); - if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4) + if (ensoniq->ctrl & ES_1371_GPIO_OUT(4)) val = 1; ucontrol->value.integer.value[0] = val; spin_unlock_irq(&ensoniq->reg_lock); -- cgit v0.10.2 From 63a450aa4d08ccf4f53e9fa59144e746e2288319 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 19 Apr 2016 15:19:02 +0100 Subject: ASoC: da7219: Update PLL ranges and dividers to improve locking The expected MCLK frequency ranges and the associated dividers are updated to improve PLL locking in a corner scenario, with low MCLK frequency near an input divider change boundary. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 81c0708..3b1d65b 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1079,21 +1079,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, dev_err(codec->dev, "PLL input clock %d below valid range\n", da7219->mclk_rate); return -EINVAL; - } else if (da7219->mclk_rate <= 5000000) { - indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; - indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; - } else if (da7219->mclk_rate <= 10000000) { - indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; - indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7219->mclk_rate <= 20000000) { - indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; - indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7219->mclk_rate <= 40000000) { - indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; - indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 4500000) { + indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 9000000) { + indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7219->mclk_rate <= 18000000) { + indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7219->mclk_rate <= 36000000) { + indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7219->mclk_rate <= 54000000) { - indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; - indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7219->mclk_rate); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 5a787e7..ff2a2f0 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -194,11 +194,11 @@ /* DA7219_PLL_CTRL = 0x20 */ #define DA7219_PLL_INDIV_SHIFT 2 #define DA7219_PLL_INDIV_MASK (0x7 << 2) -#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) -#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) -#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) -#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) -#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_9_TO_18_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_18_TO_36_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_36_TO_54_MHZ (0x4 << 2) #define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 #define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) #define DA7219_PLL_MODE_SHIFT 6 @@ -761,11 +761,11 @@ #define DA7219_PLL_FREQ_OUT_98304 98304000 /* PLL Frequency Dividers */ -#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 -#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16 /* SRM */ #define DA7219_SRM_CHECK_RETRIES 8 -- cgit v0.10.2 From fb137ba64a6415ddf231495f6d1a82de1cd69ed0 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 19 Apr 2016 15:19:03 +0100 Subject: ASoC: da7219: Disallow unsupported 32KHz clock setting in set_dai_sysclk() The PLL function was updated to disallow 32KHz in commit 501f72e9c520 ("ASoC: da7219: Remove support for 32KHz PLL mode"), but set_dai_sysclk() was missed and still permits it. This patch resolves that discrepancy. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 3b1d65b..caea2ee 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1025,7 +1025,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) return 0; - if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + if ((freq < 2000000) || (freq > 54000000)) { dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; -- cgit v0.10.2 From b6bf3289bc3c1d8df9f37c2f4f8450cc677fb286 Mon Sep 17 00:00:00 2001 From: Stephen Boyd Date: Tue, 19 Apr 2016 18:05:04 -0700 Subject: ASoC: ak4642: Remove CLK_IS_ROOT This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Signed-off-by: Stephen Boyd Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c2..1ee8506 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); -- cgit v0.10.2 From 2f0ad49104cbb19db24442af736614659363d2ab Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 19 Apr 2016 13:12:35 +0800 Subject: ASoC: Change DAI link's be_id to a generic id The generic ID can be used by topology: - Toplogy can create FE links and set their ID, machine drivers will be notified and check this ID for machine-specific init. - Toplogy can use the ID to find existing BE & CC links and further configure them. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..ef25e86 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1002,7 +1002,7 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - int be_id; /* optional ID for machine driver BE identification */ + int id; /* optional ID for machine driver link identification */ const struct snd_soc_pcm_stream *params; unsigned int num_params; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3f8a1e1..7486a00 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e7..88efb62 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1c95ccc..35f591e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index e609f08..6260df6 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2a6f808..0618a7f 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2e5347f..df9d254 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 2255857..863f1d5 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b7..9cc9240 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -456,7 +456,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -472,7 +472,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -489,7 +489,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -501,7 +501,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -512,7 +512,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99..53380b2 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -505,7 +505,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -540,7 +540,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -552,7 +552,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -563,7 +563,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -574,7 +574,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a..9e39fc1 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -375,7 +375,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -393,7 +393,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "dmic01", - .be_id = 1, + .id = 1, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -405,7 +405,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp1", - .be_id = 2, + .id = 2, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -416,7 +416,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp2", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -427,7 +427,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp3", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", -- cgit v0.10.2 From 305e9020f09d28560373c0112682e6fd11e909f6 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 19 Apr 2016 13:12:25 +0800 Subject: ASoC: Export snd_soc_find_dai() This API can be used by topology to find an existing BE dai by name and further configure it. Topology will also check DAI ID to avoid wrong match. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..7687e2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, int snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv); +struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc); + #include #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..07663de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,7 +930,7 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_dai *snd_soc_find_dai( +struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { struct snd_soc_component *component; @@ -959,6 +959,7 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_find_dai); static bool soc_is_dai_link_bound(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) -- cgit v0.10.2 From 8e42db1eaab6c2558dbc2e6c1428730df0a295f4 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 21 Apr 2016 14:04:14 +0100 Subject: ASoC: arizona: Prefer lower FRATIO in pseudo-fractional mode When setting up an FLL in pseudo-fractional mode it is preferred to use a lower FRATIO if possible to give a higher reference clock frequency. This patch swaps the two loops in arizona_calc_fratio() so that lower FRATIOs are tried first. The decrementing loop is also changed to start from init_ratio because the original settings might already give a fractional value for N.K Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d8a6823..0caecc6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2037,7 +2037,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll, init_ratio, Fref, refdiv); while (div <= ARIZONA_FLL_MAX_REFDIV) { - for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + /* start from init_ratio because this may already give a + * fractional N.K + */ + for (ratio = init_ratio; ratio > 0; ratio--) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); + return ratio; + } + } + + for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / (fll->vco_mult * ratio) < Fref) { @@ -2063,17 +2077,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (target % (ratio * Fref)) { - cfg->refdiv = refdiv; - cfg->fratio = ratio - 1; - arizona_fll_dbg(fll, - "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", - Fref, refdiv, div, ratio); - return ratio; - } - } - div *= 2; Fref /= 2; refdiv++; -- cgit v0.10.2 From 09305da97c7808b900985526aa9198233f32fb37 Mon Sep 17 00:00:00 2001 From: Shreyas NC Date: Thu, 21 Apr 2016 11:45:22 +0530 Subject: ASoC: Intel: Skylake: Use UUID in binary format To avoid complex string manipulations with UUID in canonical form, use UUID in binary format. Signed-off-by: Shreyas NC Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index ff31e66..deabe73 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -118,7 +118,7 @@ struct skl_dsp_fw_ops { int (*set_state_D0)(struct sst_dsp *ctx); int (*set_state_D3)(struct sst_dsp *ctx); unsigned int (*get_fw_errcode)(struct sst_dsp *ctx); - int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name); + int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name); int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id); }; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 348a734..bec4a7c 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -20,6 +20,7 @@ #include #include #include +#include #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" @@ -304,14 +305,16 @@ static int skl_transfer_module(struct sst_dsp *ctx, return ret; } -static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) +static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) { struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ + uuid_le *uuid_mod; - snprintf(mod_name, sizeof(mod_name), "%s%s%s", - "intel/dsp_fw_", guid, ".bin"); + uuid_mod = (uuid_le *)guid; + snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", + "intel/dsp_fw_", uuid_mod, ".bin"); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e7..4f27d82 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1550,6 +1550,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, return -ENOMEM; w->priv = mconfig; + memcpy(&mconfig->guid, &dfw_config->uuid, 16); + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; @@ -1579,10 +1581,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->time_slot = dfw_config->time_slot; mconfig->formats_config.caps_size = dfw_config->caps.caps_size; - if (dfw_config->is_loadable) - memcpy(mconfig->guid, dfw_config->uuid, - ARRAY_SIZE(dfw_config->uuid)); - mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) * sizeof(*mconfig->m_in_pin), GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index de3c401..22c913d 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -281,7 +281,7 @@ enum skl_module_state { }; struct skl_module_cfg { - char guid[SKL_UUID_STR_SZ]; + u8 guid[16]; struct skl_module_inst_id id; u8 domain; bool homogenous_inputs; diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h index 1db88a6..a32e5e9 100644 --- a/sound/soc/intel/skylake/skl-tplg-interface.h +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -181,7 +181,7 @@ struct skl_dfw_pipe { } __packed; struct skl_dfw_module { - char uuid[SKL_UUID_STR_SZ]; + u8 uuid[16]; u16 module_id; u16 instance_id; -- cgit v0.10.2 From 81151cfb6bfe69f1c5a52b795eb005226a322c9e Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Wed, 20 Apr 2016 10:59:58 +0200 Subject: ASoC: hdmi-codec: Add ELD control ALSA doesn't know about all the different compressed audio formats, so there is no interface to let userspace enumerate the formats that are supported by the connected sink. Exporting the raw ELD bytes to userspace allows an application to select the appropriate audio format depending on the current capabilities of the connected HDMI sink device. Usually userspace then just pretends to ALSA that the data is in one of the raw 16-bit PCM audio formats and relies on the IEC controls to tell the sink how to interpret the data. Signed-off-by: Philipp Zabel Reviewed-by: Jyri Sarha Tested-by: Jyri Sarha Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b46b8ed..c78333b 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -47,6 +47,42 @@ enum { DAI_ID_SPDIF, }; +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = sizeof(hcp->eld); + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + mutex_lock(&hcp->eld_lock); + memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld)); + mutex_unlock(&hcp->eld_lock); + + return 0; +} + +static const struct snd_kcontrol_new hdmi_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, + }, +}; + static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -312,6 +348,8 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { }; static struct snd_soc_codec_driver hdmi_codec = { + .controls = hdmi_controls, + .num_controls = ARRAY_SIZE(hdmi_controls), .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, -- cgit v0.10.2 From db71336b9eec22c21cef65c90cea49130c464994 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Fri, 22 Apr 2016 10:40:11 +0200 Subject: ASoC: hdmi-codec: Add ELD control Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index c78333b..8e36e88 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -65,9 +65,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); - mutex_lock(&hcp->eld_lock); memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld)); - mutex_unlock(&hcp->eld_lock); return 0; } -- cgit v0.10.2 From fba0d7066524ab7b8ccf60e7e95981d10ed008b0 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 22 Apr 2016 09:03:53 +0530 Subject: ASoC: Intel: Atom: fix boot warning Users have reported seeing this false warning on atom driver [ 5.647469] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.661612] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.661646] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.661681] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.661708] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.661738] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.661771] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.661807] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! This is caused when check for control is not being associated with a dapm widget, but the check is wrong as the else case maybe triggered when widget is not powered up, so we should check if widget is associated before printing this message. Tested-by: Sandeep Tayal Tested-by: Pierre-Louis Bossart Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index b97e6ad..98720a9 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol if (e->w && e->w->power) ret = sst_send_slot_map(drv); - else + else if (!e->w) dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", kcontrol->id.name); return ret; -- cgit v0.10.2 From fbb88b5ca1dc84416fc1fec34773948b6780492c Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 22 Apr 2016 12:25:33 +0800 Subject: ASoC: Add kerneldoc comments for snd_soc_find_dai snd_soc_find_dai() has been exported and so add the kerneldoc comments for it. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07663de..16369ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,6 +930,17 @@ static struct snd_soc_component *soc_find_component( return NULL; } +/** + * snd_soc_find_dai - Find a registered DAI + * + * @dlc: name of the DAI and optional component info to match + * + * This function will search all regsitered components and their DAIs to + * find the DAI of the same name. The component's of_node and name + * should also match if being specified. + * + * Return: pointer of DAI, or NULL if not found. + */ struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { -- cgit v0.10.2 From ae48a35c408732413880d0ac0d6467baa5b3d68a Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Fri, 22 Apr 2016 14:16:26 +0100 Subject: ASoC: da7218: Update PLL ranges and dividers to improve locking The expected MCLK frequency ranges and the associated dividers are updated to improve PLL locking in a corner scenario, with low MCLK frequency near an input divider change boundary. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 93575f2..99ce23e 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ if (da7218->mclk_rate == 32768) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; } else if (da7218->mclk_rate < 2000000) { dev_err(codec->dev, "PLL input clock %d below valid range\n", da7218->mclk_rate); return -EINVAL; - } else if (da7218->mclk_rate <= 5000000) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 10000000) { - indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 20000000) { - indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; - indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7218->mclk_rate <= 40000000) { - indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; - indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 4500000) { + indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7218->mclk_rate <= 9000000) { + indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7218->mclk_rate <= 18000000) { + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7218->mclk_rate <= 36000000) { + indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7218->mclk_rate <= 54000000) { - indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; - indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7218->mclk_rate); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h index c2c5904..477cd37 100644 --- a/sound/soc/codecs/da7218.h +++ b/sound/soc/codecs/da7218.h @@ -876,15 +876,11 @@ /* DA7218_PLL_CTRL = 0x91 */ #define DA7218_PLL_INDIV_SHIFT 0 #define DA7218_PLL_INDIV_MASK (0x7 << 0) -#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) -#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) -#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) -#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) -#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) -#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 -#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0) #define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 #define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) #define DA7218_PLL_MODE_SHIFT 6 @@ -1336,6 +1332,13 @@ #define DA7218_PLL_FREQ_OUT_90316 90316800 #define DA7218_PLL_FREQ_OUT_98304 98304000 +/* PLL Frequency Dividers */ +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16 + /* ALC Calibration */ #define DA7218_ALC_CALIB_DELAY_MIN 2500 #define DA7218_ALC_CALIB_DELAY_MAX 5000 -- cgit v0.10.2 From b610386c8afba397238329c50c45a3abc79ba45f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 31 Mar 2016 08:47:09 +0900 Subject: ALSA: firewire-tascam: deleyed registration of sound card When some tascam units are connected sequentially, userspace applications are involved at bus-reset state on IEEE 1394 bus. In the state, any communications can be canceled. Therefore, sound card registration should be delayed till the bus gets calm. This commit achieves it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index e281c33..9dc93a7 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -85,10 +85,8 @@ static int identify_model(struct snd_tscm *tscm) return 0; } -static void tscm_card_free(struct snd_card *card) +static void tscm_free(struct snd_tscm *tscm) { - struct snd_tscm *tscm = card->private_data; - snd_tscm_transaction_unregister(tscm); snd_tscm_stream_destroy_duplex(tscm); @@ -97,44 +95,36 @@ static void tscm_card_free(struct snd_card *card) mutex_destroy(&tscm->mutex); } -static int snd_tscm_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void tscm_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_tscm *tscm; + tscm_free(card->private_data); +} + +static void do_registration(struct work_struct *work) +{ + struct snd_tscm *tscm = container_of(work, struct snd_tscm, dwork.work); int err; - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_tscm), &card); + err = snd_card_new(&tscm->unit->device, -1, NULL, THIS_MODULE, 0, + &tscm->card); if (err < 0) - return err; - card->private_free = tscm_card_free; - - /* initialize myself */ - tscm = card->private_data; - tscm->card = card; - tscm->unit = fw_unit_get(unit); - - mutex_init(&tscm->mutex); - spin_lock_init(&tscm->lock); - init_waitqueue_head(&tscm->hwdep_wait); + return; err = identify_model(tscm); if (err < 0) goto error; - snd_tscm_proc_init(tscm); - - err = snd_tscm_stream_init_duplex(tscm); + err = snd_tscm_transaction_register(tscm); if (err < 0) goto error; - err = snd_tscm_create_pcm_devices(tscm); + err = snd_tscm_stream_init_duplex(tscm); if (err < 0) goto error; - err = snd_tscm_transaction_register(tscm); + snd_tscm_proc_init(tscm); + + err = snd_tscm_create_pcm_devices(tscm); if (err < 0) goto error; @@ -146,35 +136,91 @@ static int snd_tscm_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(tscm->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, tscm); + /* + * After registered, tscm instance can be released corresponding to + * releasing the sound card instance. + */ + tscm->card->private_free = tscm_card_free; + tscm->card->private_data = tscm; + tscm->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_tscm_transaction_unregister(tscm); + snd_tscm_stream_destroy_duplex(tscm); + snd_card_free(tscm->card); + dev_info(&tscm->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_tscm_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_tscm *tscm; + + /* Allocate this independent of sound card instance. */ + tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL); + if (tscm == NULL) + return -ENOMEM; + + /* initialize myself */ + tscm->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, tscm); + + mutex_init(&tscm->mutex); + spin_lock_init(&tscm->lock); + init_waitqueue_head(&tscm->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&tscm->dwork, do_registration); + snd_fw_schedule_registration(unit, &tscm->dwork); + + return 0; } static void snd_tscm_update(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!tscm->registered) + snd_fw_schedule_registration(unit, &tscm->dwork); + snd_tscm_transaction_reregister(tscm); - mutex_lock(&tscm->mutex); - snd_tscm_stream_update_duplex(tscm); - mutex_unlock(&tscm->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (tscm->registered) { + mutex_lock(&tscm->mutex); + snd_tscm_stream_update_duplex(tscm); + mutex_unlock(&tscm->mutex); + } } static void snd_tscm_remove(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(tscm->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&tscm->dwork); + + if (tscm->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(tscm->card); + } else { + /* Don't forget this case. */ + tscm_free(tscm); + } } static const struct ieee1394_device_id snd_tscm_id_table[] = { diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 30ab77e..1f61011 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -51,6 +51,8 @@ struct snd_tscm { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; const struct snd_tscm_spec *spec; struct fw_iso_resources tx_resources; -- cgit v0.10.2 From 34ce71a96dcba24c71b07f1b087bd179b885784d Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Sat, 23 Apr 2016 02:58:05 +0200 Subject: ALSA: timer: remove legacy rtctimer There are no users of rtctimer left. Remove its code as this is the in-kernel user of the legacy PC RTC driver that will hopefully be removed at some point. Signed-off-by: Alexandre Belloni Signed-off-by: Takashi Iwai diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 67bf49d..609cadb 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -672,7 +672,7 @@ enum { /* global timers (device member) */ #define SNDRV_TIMER_GLOBAL_SYSTEM 0 -#define SNDRV_TIMER_GLOBAL_RTC 1 +#define SNDRV_TIMER_GLOBAL_RTC 1 /* unused */ #define SNDRV_TIMER_GLOBAL_HPET 2 #define SNDRV_TIMER_GLOBAL_HRTIMER 3 diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6d12ca9..9749f9e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT Say Y here to use the HR-timer backend as the default sequencer timer. -config SND_RTCTIMER - tristate "RTC Timer support" - depends on RTC - select SND_TIMER - help - Say Y here to enable RTC timer support for ALSA. ALSA uses - the RTC timer as a precise timing source and maps the RTC - timer to ALSA's timer interface. The ALSA sequencer code also - can use this timing source. - - To compile this driver as a module, choose M here: the module - will be called snd-rtctimer. - - Note that this option is exclusive with the new RTC drivers - (CONFIG_RTC_CLASS) since this requires the old API. - -config SND_SEQ_RTCTIMER_DEFAULT - bool "Use RTC as default sequencer timer" - depends on SND_RTCTIMER && SND_SEQUENCER - depends on !SND_SEQ_HRTIMER_DEFAULT - default y - help - Say Y here to use the RTC timer as the default sequencer - timer. This is strongly recommended because it ensures - precise MIDI timing even when the system timer runs at less - than 1000 Hz. - - If in doubt, say Y. - config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" help diff --git a/sound/core/Makefile b/sound/core/Makefile index 48ab4b8..e85d9dd 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o -obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d1..0000000 --- a/sound/core/rtctimer.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * RTC based high-frequency timer - * - * Copyright (C) 2000 Takashi Iwai - * based on rtctimer.c by Steve Ratcliffe - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#if IS_ENABLED(CONFIG_RTC) - -#include - -#define RTC_FREQ 1024 /* default frequency */ -#define NANO_SEC 1000000000L /* 10^9 in sec */ - -/* - * prototypes - */ -static int rtctimer_open(struct snd_timer *t); -static int rtctimer_close(struct snd_timer *t); -static int rtctimer_start(struct snd_timer *t); -static int rtctimer_stop(struct snd_timer *t); - - -/* - * The hardware dependent description for this timer. - */ -static struct snd_timer_hardware rtc_hw = { - .flags = SNDRV_TIMER_HW_AUTO | - SNDRV_TIMER_HW_FIRST | - SNDRV_TIMER_HW_TASKLET, - .ticks = 100000000L, /* FIXME: XXX */ - .open = rtctimer_open, - .close = rtctimer_close, - .start = rtctimer_start, - .stop = rtctimer_stop, -}; - -static int rtctimer_freq = RTC_FREQ; /* frequency */ -static struct snd_timer *rtctimer; -static struct tasklet_struct rtc_tasklet; -static rtc_task_t rtc_task; - - -static int -rtctimer_open(struct snd_timer *t) -{ - int err; - - err = rtc_register(&rtc_task); - if (err < 0) - return err; - t->private_data = &rtc_task; - return 0; -} - -static int -rtctimer_close(struct snd_timer *t) -{ - rtc_task_t *rtc = t->private_data; - if (rtc) { - rtc_unregister(rtc); - tasklet_kill(&rtc_tasklet); - t->private_data = NULL; - } - return 0; -} - -static int -rtctimer_start(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq); - rtc_control(rtc, RTC_PIE_ON, 0); - return 0; -} - -static int -rtctimer_stop(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_PIE_OFF, 0); - return 0; -} - -static void rtctimer_tasklet(unsigned long data) -{ - snd_timer_interrupt((struct snd_timer *)data, 1); -} - -/* - * interrupt - */ -static void rtctimer_interrupt(void *private_data) -{ - tasklet_schedule(private_data); -} - - -/* - * ENTRY functions - */ -static int __init rtctimer_init(void) -{ - int err; - struct snd_timer *timer; - - if (rtctimer_freq < 2 || rtctimer_freq > 8192 || - !is_power_of_2(rtctimer_freq)) { - pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq); - return -EINVAL; - } - - /* Create a new timer and set up the fields */ - err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer); - if (err < 0) - return err; - - timer->module = THIS_MODULE; - strcpy(timer->name, "RTC timer"); - timer->hw = rtc_hw; - timer->hw.resolution = NANO_SEC / rtctimer_freq; - - tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer); - - /* set up RTC callback */ - rtc_task.func = rtctimer_interrupt; - rtc_task.private_data = &rtc_tasklet; - - err = snd_timer_global_register(timer); - if (err < 0) { - snd_timer_global_free(timer); - return err; - } - rtctimer = timer; /* remember this */ - - return 0; -} - -static void __exit rtctimer_exit(void) -{ - if (rtctimer) { - snd_timer_global_free(rtctimer); - rtctimer = NULL; - } -} - - -/* - * exported stuff - */ -module_init(rtctimer_init) -module_exit(rtctimer_exit) - -module_param(rtctimer_freq, int, 0444); -MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz"); - -MODULE_LICENSE("GPL"); - -MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); - -#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7e0aabb..639544b 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; int seq_default_timer_device = #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT SNDRV_TIMER_GLOBAL_HRTIMER -#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) - SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6469bed..0cfc028 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -37,8 +37,6 @@ #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif IS_ENABLED(CONFIG_SND_RTCTIMER) -#define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 #endif -- cgit v0.10.2 From 8d84c1973b69621bcb89d5d8a69699e8347a3d90 Mon Sep 17 00:00:00 2001 From: Eric Engestrom Date: Mon, 25 Apr 2016 07:37:02 +0100 Subject: ALSA: doc: fix spelling mistakes Signed-off-by: Eric Engestrom Acked-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492..81476b4 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 6faab48..c45bd79 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), -If you dont want the mixer elements prefixed with the name of the mixer widget, +If you don't want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index ff88f52..f3f28b7 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -63,7 +63,7 @@ multiple re-usable component drivers :- and any audio DSP drivers for that platform. * Machine class driver: The machine driver class acts as the glue that - decribes and binds the other component drivers together to form an ALSA + describes and binds the other component drivers together to form an ALSA "sound card device". It handles any machine specific controls and machine level audio events (e.g. turning on an amp at start of playback). -- cgit v0.10.2 From a34b027dca5ea840fbc84121db66488375acfdea Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Mon, 25 Apr 2016 13:39:38 +0000 Subject: ASoC: bcm2835: add 24bit support This adds 24 bit support to the I2S driver of the BCM2835 Code ported from bcm2708-i2s driver in Raspberry Pi tree. Signed-off-by: Florian Meier Signed-off-by: Matthias Reichl Signed-off-by: Martin Sperl Signed-off-by: Mark Brown diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 1c1f221..d2663e7 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: data_length = 16; break; + case SNDRV_PCM_FORMAT_S24_LE: + data_length = 24; + break; case SNDRV_PCM_FORMAT_S32_LE: data_length = 32; break; @@ -279,7 +282,7 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Setup the frame format */ format = BCM2835_I2S_CHEN; - if (data_length > 24) + if (data_length >= 24) format |= BCM2835_I2S_CHWEX; format |= BCM2835_I2S_CHWID((data_length-8)&0xf); @@ -570,6 +573,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .capture = { @@ -577,6 +581,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .ops = &bcm2835_i2s_dai_ops, -- cgit v0.10.2 From 60507fe191f524e82986fa737e5b27b4d3ad9289 Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Mon, 25 Apr 2016 13:39:39 +0000 Subject: ASoC: bcm2835: setup clock only if CPU is clock master We only need to enable the clock if we are a clock master. Code ported from bcm2708-i2s driver in Raspberry Pi tree. Original work by Zoltan Szenczi. Signed-off-by: Matthias Reichl Signed-off-by: Martin Sperl Signed-off-by: Mark Brown diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index d2663e7..a0026e2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -276,8 +276,15 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* otherwise calculate a fitting block ratio */ bclk_ratio = 2 * data_length; - /* set target clock rate*/ - clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + /* Clock should only be set up here if CPU is clock master */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + break; + default: + break; + } /* Setup the frame format */ format = BCM2835_I2S_CHEN; -- cgit v0.10.2 From d2c5cf88d5282de258f4eb6ab40040b80a075cd8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 24 Apr 2016 22:52:18 +0200 Subject: ALSA: hrtimer: Handle start/stop more properly This patch tries to address the still remaining issues in ALSA hrtimer driver: - Spurious use-after-free was detected in hrtimer callback - Incorrect rescheduling due to delayed start - WARN_ON() is triggered in hrtimer_forward() invoked in hrtimer callback The first issue happens only when the new timer is scheduled even while hrtimer is being closed. It's related with the second and third items; since ALSA timer core invokes hw.start callback during hrtimer interrupt, this may result in the explicit call of hrtimer_start(). Also, the similar problem is seen for the stop; ALSA timer core invokes hw.stop callback even in the hrtimer handler, too. Since we must not call the synced hrtimer_cancel() in such a context, it's just a hrtimer_try_to_cancel() call that doesn't properly work. Another culprit of the second and third items is the call of hrtimer_forward_now() before snd_timer_interrupt(). The timer->stick value may change during snd_timer_interrupt() call, but this possibility is ignored completely. For covering these subtle and messy issues, the following changes have been done in this patch: - A new flag, in_callback, is introduced in the private data to indicate that the hrtimer handler is being processed. - Both start and stop callbacks skip when called from (during) in_callback flag. - The hrtimer handler returns properly HRTIMER_RESTART and NORESTART depending on the running state now. - The hrtimer handler reprograms the expiry properly after snd_timer_interrupt() call, instead of before. - The close callback clears running flag and sets in_callback flag to block any further start/stop calls. Signed-off-by: Takashi Iwai diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 656d9a9..e2f2702 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -38,37 +38,53 @@ static unsigned int resolution; struct snd_hrtimer { struct snd_timer *timer; struct hrtimer hrt; - atomic_t running; + bool in_callback; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; - unsigned long oruns; - - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - - oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks * oruns); + ktime_t delta; + unsigned long ticks; + enum hrtimer_restart ret = HRTIMER_NORESTART; + + spin_lock(&t->lock); + if (!t->running) + goto out; /* fast path */ + stime->in_callback = true; + ticks = t->sticks; + spin_unlock(&t->lock); + + /* calculate the drift */ + delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt)); + if (delta.tv64 > 0) + ticks += ktime_divns(delta, ticks * resolution); + + snd_timer_interrupt(stime->timer, ticks); + + spin_lock(&t->lock); + if (t->running) { + hrtimer_add_expires_ns(hrt, t->sticks * resolution); + ret = HRTIMER_RESTART; + } - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - return HRTIMER_RESTART; + stime->in_callback = false; + out: + spin_unlock(&t->lock); + return ret; } static int snd_hrtimer_open(struct snd_timer *t) { struct snd_hrtimer *stime; - stime = kmalloc(sizeof(*stime), GFP_KERNEL); + stime = kzalloc(sizeof(*stime), GFP_KERNEL); if (!stime) return -ENOMEM; hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; stime->hrt.function = snd_hrtimer_callback; - atomic_set(&stime->running, 0); t->private_data = stime; return 0; } @@ -78,6 +94,11 @@ static int snd_hrtimer_close(struct snd_timer *t) struct snd_hrtimer *stime = t->private_data; if (stime) { + spin_lock_irq(&t->lock); + t->running = 0; /* just to be sure */ + stime->in_callback = 1; /* skip start/stop */ + spin_unlock_irq(&t->lock); + hrtimer_cancel(&stime->hrt); kfree(stime); t->private_data = NULL; @@ -89,18 +110,19 @@ static int snd_hrtimer_start(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); - hrtimer_try_to_cancel(&stime->hrt); + if (stime->in_callback) + return 0; hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); - atomic_set(&stime->running, 1); return 0; } static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); + + if (stime->in_callback) + return 0; hrtimer_try_to_cancel(&stime->hrt); return 0; } -- cgit v0.10.2 From 58a8738cfcdec389b3764a636303f97b57f85193 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Mar 2016 21:03:09 +0100 Subject: ALSA: au88x0: Fix overlapped PCM pointer au88x0 hardware seems returning the current pointer at the buffer boundary instead of going back to zero. This results in spewing warnings from PCM core. This patch corrects the return value from the pointer callback within the proper value range, just returning zero if the position is equal or above the buffer size. Signed-off-by: Takashi Iwai diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index a6d6d8d..df5741a 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -432,7 +432,10 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr #endif //printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr); spin_unlock(&chip->lock); - return (bytes_to_frames(substream->runtime, current_ptr)); + current_ptr = bytes_to_frames(substream->runtime, current_ptr); + if (current_ptr >= substream->runtime->buffer_size) + current_ptr = 0; + return current_ptr; } /* operators */ -- cgit v0.10.2 From de06f22f717b30641229036439b804ae79a7ad4d Mon Sep 17 00:00:00 2001 From: Javier Martinez Canillas Date: Mon, 25 Apr 2016 19:30:39 -0400 Subject: ASoC: cs42l56: Use IS_ENABLED() instead of checking for built-in or module The IS_ENABLED() macro checks if a Kconfig symbol has been enabled either built-in or as a module, use that macro instead of open coding the same. Signed-off-by: Javier Martinez Canillas Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cd5f76..eec1ff8 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -56,7 +56,7 @@ struct cs42l56_private { u8 iface; u8 iface_fmt; u8 iface_inv; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -- cgit v0.10.2 From 2ab8e744a437d39619b323d7303fa2e6513274b2 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 26 Apr 2016 17:06:20 +0100 Subject: ASoC: arizona: No need to update_bits when writing AEC clock control The bits in the ARIZONA_CLOCK_CONTROL register only respond to writes of a '1', a write of '0' is ignored. So there's no need to use update_bits. We can do a simple write to set bits. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0caecc6..0239639 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1124,7 +1124,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int mask = 0x3 << w->shift; unsigned int val; switch (event) { @@ -1138,7 +1137,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, return 0; } - snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val); + snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val); return 0; } -- cgit v0.10.2 From 80833ff0eea693d8e0c3305a869159a64141fdad Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Wed, 27 Apr 2016 09:06:49 +0200 Subject: ASoC: atmel_ssc_dai: read DSP mode A data on rising edges of bclk Signed-off-by: Peter Rosin Signed-off-by: Mark Brown diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 2768970..1267e1a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); -- cgit v0.10.2 From 66225e98b985047ef214632413cc404a6341c960 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 27 Apr 2016 14:58:27 +0100 Subject: ASoC: wm_adsp: free memory when unloaded or closed The patch adds a wm_adsp2_remove() function to ensure that memory is freed when the driver is unloaded or shut down. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb1..5f8727a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -944,6 +944,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -2340,6 +2347,19 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57..feb61e2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, -- cgit v0.10.2 From 401cf1466a59139ec1805e2171d43a32be92f89c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 27 Apr 2016 14:58:28 +0100 Subject: ASoC: arizona: call wm_adsp2_remove when codec driver is removed Ensure that the wm_adsp driver cleans up when the codec driver is removed. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 576087b..2931378 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1238,10 +1238,15 @@ static int cs47l24_probe(struct platform_device *pdev) static int cs47l24_remove(struct platform_device *pdev) { + struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&cs47l24->core.adsp[1]); + wm_adsp2_remove(&cs47l24->core.adsp[2]); + return 0; } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a8b3e3f..7a539e0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2093,10 +2093,14 @@ static int wm5102_probe(struct platform_device *pdev) static int wm5102_remove(struct platform_device *pdev) { + struct wm5102_priv *wm5102 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&wm5102->core.adsp[0]); + return 0; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 83ba70f..dd87af1 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2435,10 +2435,16 @@ static int wm5110_probe(struct platform_device *pdev) static int wm5110_remove(struct platform_device *pdev) { + struct wm5110_priv *wm5110 = platform_get_drvdata(pdev); + int i; + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp2_remove(&wm5110->core.adsp[i]); + return 0; } -- cgit v0.10.2 From 56574d541f93cf8c9449f9ecadc83d97323cfcec Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 27 Apr 2016 14:58:29 +0100 Subject: ASoC: wm_adsp: factor out freeing of alg regions Add a function to delete and free the contents of the alg_regions list. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5f8727a..8cde7bb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1571,6 +1571,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp, return alg_region; } +static void wm_adsp_free_alg_regions(struct wm_adsp *dsp) +{ + struct wm_adsp_alg_region *alg_region; + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } +} + static int wm_adsp1_setup_algs(struct wm_adsp *dsp) { struct wmfw_adsp1_id_hdr adsp1_id; @@ -2001,7 +2014,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; unsigned int val; @@ -2081,13 +2093,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + + wm_adsp_free_alg_regions(dsp); break; default: @@ -2229,7 +2236,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; @@ -2276,13 +2282,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + wm_adsp_free_alg_regions(dsp); if (wm_adsp_fw[dsp->fw].num_caps != 0) wm_adsp_buffer_free(dsp); -- cgit v0.10.2 From 73fe01cfb3babff01748a9fbc95cc3ea2079cc7f Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Wed, 27 Apr 2016 15:26:51 +0200 Subject: ASoC: dmaengine_pcm: Add support for packed transfers dmaengine_pcm currently only supports setups where FIFO reads/writes correspond to exactly one sample, eg 16-bit sample data is transferred via 16-bit FIFO accesses, 32-bit data via 32-bit accesses. This patch adds support for setups with fixed width FIFOs where multiple samples are packed into a larger word. For example setups with a 32-bit wide FIFO register that expect 16-bit sample transfers to be done with the left+right sample data packed into a 32-bit word. Support for packed transfers is controlled via the SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags If this flag is set dmaengine_pcm doesn't put any restriction on the supported formats and sets the DMA transfer width to undefined. This means control over the constraints is now transferred to the DAI driver and it's responsible to provide proper configuration and check for possible corner cases that aren't handled by the ALSA core. Signed-off-by: Matthias Reichl Acked-by: Lars-Peter Clausen Tested-by: Martin Sperl Signed-off-by: Mark Brown diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f86ef5e..67be244 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, void *filter_data); struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); +/* + * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word. + * If this flag is set the dmaengine driver won't put any restriction on + * the supported sample formats and set the DMA transfer size to undefined. + * The DAI driver is responsible to disable any unsupported formats in it's + * configuration and catch corner cases that are not already handled in + * the ALSA core. + */ +#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0) + /** * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data * @addr: Address of the DAI data source or destination register. @@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * requesting the DMA channel. * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes + * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data { void *filter_data; const char *chan_name; unsigned int fifo_size; + unsigned int flags; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166..8eb58c7 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906..6cef397 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } -- cgit v0.10.2 From beff053c0ef6983897e3481169292e6435ef0a2d Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Wed, 27 Apr 2016 15:26:52 +0200 Subject: ASoC: bcm2835: Add S16_LE support via packed DMA transfers The bcm2835-i2s driver already has support for the S16_LE format but that format hasn't been made available because dmaengine_pcm didn't support packed data transfers. bcm2835-i2s needs 16-bit left+right channel data to be packed into a 32-bit word, the FIFO register is 32-bit only and doesn't support 16-bit access. Now that dmaengine_pcm supports packed transfers the format can be made available by setting the SND_DMAENGINE_PCM_DAI_FLAG_PACK flag. No further configuration is necessary: - snd_dmaengine_dai_dma_data.addr_width is already set to DMA_SLAVE_BUSWIDTH_4_BYTES to force 32-bit DMA transfers - dmaengine_pcm will pick up the S16_LE format from the DAI configuration and make it available since it's no longer masked out due to the PACK flag. - there are no further corner cases to catch in hw_params, since the channel count is fixed at 2 we always have two 16-bit stereo samples that can be transferred via 32-bit DMA Signed-off-by: Matthias Reichl Tested-by: Martin Sperl Signed-off-by: Mark Brown diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index a0026e2..6ba2049 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -690,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + /* + * Set the PACK flag to enable S16_LE support (2 S16_LE values + * packed into 32-bit transfers). + */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + /* BCLK ratio - use default */ dev->bclk_ratio = 0; -- cgit v0.10.2 From 9fc7c862e78ba8bec142935673227f2463aa05a5 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:29 +0530 Subject: ALSA: hda - add helper to get channels from cap bits This helper is copied from legacy hda driver. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/sound/hda/local.h b/sound/hda/local.h index d692f41..0d5bb15 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps) return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; } +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = (chans + 1) * 2; + + return chans; +} + extern const struct attribute_group *hdac_dev_attr_groups[]; int hda_widget_sysfs_init(struct hdac_device *codec); void hda_widget_sysfs_exit(struct hdac_device *codec); -- cgit v0.10.2 From b7756edeb7d03b675e10b4862dccc8deb4b0ca17 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:28 +0530 Subject: ASoC: hdac_hdmi: parse eld for channel map capability This patch parses ELD speaker allocation data block to find sink's chmap capability. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 26f9459..64ffe93 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -60,11 +60,17 @@ struct hdac_hdmi_cvt { struct hdac_hdmi_cvt_params params; }; +/* Currently only spk_alloc, more to be added */ +struct hdac_hdmi_parsed_eld { + u8 spk_alloc; +}; + struct hdac_hdmi_eld { bool monitor_present; bool eld_valid; int eld_size; char eld_buffer[ELD_MAX_SIZE]; + struct hdac_hdmi_parsed_eld info; }; struct hdac_hdmi_pin { @@ -1008,6 +1014,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) return hdac_hdmi_query_cvt_params(&edev->hdac, cvt); } +static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev, + struct hdac_hdmi_pin *pin) +{ + pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER]; +} + static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; @@ -1065,6 +1077,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) snd_jack_report(pcm->jack, SND_JACK_AVOUT); } + hdac_hdmi_parse_eld(edev, pin); print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET, pin->eld.eld_buffer, pin->eld.eld_size); -- cgit v0.10.2 From bcced704788312360c0413d13b11611ae00a91c8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:30 +0530 Subject: ASoC: hdac_hdmi: Add multichannel support To support multichannel hdac hdmi driver registers with HDA channel map framework. Channel count and channel slot verbs are programmed by using the chmap helpers/ops. The channel allocation is then programmed in the audio infoframe as per CEA spec. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 64ffe93..034593b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -29,6 +29,7 @@ #include #include #include +#include #include "../../hda/local.h" #include "hdac_hdmi.h" @@ -82,6 +83,10 @@ struct hdac_hdmi_pin { struct hdac_ext_device *edev; int repoll_count; struct delayed_work work; + struct mutex lock; + bool chmap_set; + unsigned char chmap[8]; /* ALSA API channel-map */ + int channels; /* current number of channels */ }; struct hdac_hdmi_pcm { @@ -106,6 +111,7 @@ struct hdac_hdmi_priv { int num_pin; int num_cvt; struct mutex pin_mutex; + struct hdac_chmap chmap; }; static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) @@ -284,26 +290,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, int i; const u8 *eld_buf; u8 conn_type; - int channels = 2; + int channels, ca; list_for_each_entry(pin, &hdmi->pin_list, head) { if (pin->nid == pin_nid) break; } + ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc, + pin->channels, pin->chmap_set, true, pin->chmap); + + channels = snd_hdac_get_active_channels(ca); + hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels); + + snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, + pin->channels, pin->chmap, pin->chmap_set); + eld_buf = pin->eld.eld_buffer; conn_type = drm_eld_get_conn_type(eld_buf); - /* setup channel count */ - snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, channels - 1); - switch (conn_type) { case DRM_ELD_CONN_TYPE_HDMI: hdmi_audio_infoframe_init(&frame); - /* Default stereo for now */ frame.channels = channels; + frame.channel_allocation = ca; ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer)); if (ret < 0) @@ -317,7 +328,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, dp_ai.len = 0x1b; dp_ai.ver = 0x11 << 2; dp_ai.CC02_CT47 = channels - 1; - dp_ai.CA = 0; + dp_ai.CA = ca; dip = (u8 *)&dp_ai; break; @@ -376,17 +387,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream, struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_priv *hdmi = hdac->private_data; struct hdac_hdmi_dai_pin_map *dai_map; + struct hdac_hdmi_pin *pin; struct hdac_ext_dma_params *dd; int ret; dai_map = &hdmi->dai_map[dai->id]; + pin = dai_map->pin; dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream); dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n", dd->stream_tag, dd->format); + mutex_lock(&pin->lock); + pin->channels = substream->runtime->channels; + ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid, dai_map->pin->nid); + mutex_unlock(&pin->lock); if (ret < 0) return ret; @@ -646,6 +663,10 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + mutex_lock(&dai_map->pin->lock); + dai_map->pin->channels = 0; + mutex_unlock(&dai_map->pin->lock); + dai_map->pin = NULL; } } @@ -653,10 +674,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, static int hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt) { + unsigned int chans; + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; int err; - /* Only stereo supported as of now */ - cvt->params.channels_min = cvt->params.channels_max = 2; + chans = get_wcaps(hdac, cvt->nid); + chans = get_wcaps_channels(chans); + + cvt->params.channels_min = 2; + + cvt->params.channels_max = chans; + if (chans > hdmi->chmap.channels_max) + hdmi->chmap.channels_max = chans; err = snd_hdac_query_supported_pcm(hdac, cvt->nid, &cvt->params.rates, @@ -1136,6 +1166,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) hdmi->num_pin++; pin->edev = edev; + mutex_init(&pin->lock); INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld); return 0; @@ -1506,6 +1537,7 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) return -ENOMEM; edev->private_data = hdmi_priv; + snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap); dev_set_drvdata(&codec->dev, edev); -- cgit v0.10.2 From 1a10612fc3f5eb5cfa89af3f6b8181d69f79a371 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:31 +0530 Subject: ASoC: skl_rt286: Fix to support hdmi channel map support HDMI registers channel map controls per PCM. As PCMs are not registered during dai_link init callback, store the pcm ids and codec DAIs during this init callback. Register for late probe and call the jack_init API which also registers channel map in the late probe callback handler. The patch following the machine driver changes adds the channel map control in the hdac_hdmi codec driver. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a..06de802 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_rt286_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id); + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static unsigned int rates[] = { @@ -438,6 +459,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + RT286S */ static struct snd_soc_card skylake_rt286 = { .name = "skylake-rt286", @@ -451,11 +487,21 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_rt286_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_rt286.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_rt286, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); } -- cgit v0.10.2 From 0d425b4f900e4dc65bd186387dae32dbbb186e77 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:32 +0530 Subject: ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap HDMI registers channel map controls per PCM. As PCMs are not registered during dai_link init callback, store the pcm ids and codec DAIs during this init callback. Register for late probe and call the jack_init API which also registers channel map in the late probe callback handler. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b7..8ccc97c 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau8825_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -534,6 +577,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825max", @@ -547,11 +605,21 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau8825_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } -- cgit v0.10.2 From 46ed1a27fb44febb2c362fc30fcb51e8eed06e3a Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:33 +0530 Subject: ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap HDMI registers channel map controls per PCM. As PCMs are not registered during dai_link init callback, store the pcm ids and codec DAIs during this init callback. Register for late probe and call the jack_init API which also registers channel map in the late probe callback handler. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99..bde85bf 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -34,6 +34,15 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau88125_private { + struct list_head hdmi_pcm_list; +}; enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -585,6 +627,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", @@ -600,11 +657,21 @@ static struct snd_soc_card skylake_audio_card = { .codec_conf = ssm4567_codec_conf, .num_configs = ARRAY_SIZE(ssm4567_codec_conf), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau88125_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } -- cgit v0.10.2 From 2889099eb8cd0811dc2986643d46c0b62b90eeb4 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:34 +0530 Subject: ASoC: hdac_hdmi: Register chmap controls and ops With this patch, chmap controls are created and user space can set the channel map. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 034593b..0ed3975 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -114,6 +114,19 @@ struct hdac_hdmi_priv { struct hdac_chmap chmap; }; +static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, + int pcm_idx) +{ + struct hdac_hdmi_pcm *pcm; + + list_for_each_entry(pcm, &hdmi->pcm_list, head) { + if (pcm->pcm_id == pcm_idx) + return pcm; + } + + return NULL; +} + static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) { struct hdac_device *hdac = dev_to_hdac_dev(dev); @@ -664,6 +677,8 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); mutex_lock(&dai_map->pin->lock); + dai_map->pin->chmap_set = false; + memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap)); dai_map->pin->channels = 0; mutex_unlock(&dai_map->pin->lock); @@ -1386,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = { .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; +static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, + int device) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->pcm && (rtd->pcm->device == device)) + return rtd->pcm; + } + + return NULL; +} + int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) { char jack_name[NAME_SIZE]; @@ -1395,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pcm *pcm; + struct snd_pcm *snd_pcm; + int err; /* * this is a new PCM device, create new pcm and @@ -1406,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) pcm->pcm_id = device; pcm->cvt = hdmi->dai_map[dai->id].cvt; + snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device); + if (snd_pcm) { + err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); + if (err < 0) { + dev_err(&edev->hdac.dev, + "chmap control add failed with err: %d for pcm: %d\n", + err, device); + kfree(pcm); + return err; + } + } + list_add_tail(&pcm->head, &hdmi->pcm_list); sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device); @@ -1524,6 +1566,60 @@ static struct snd_soc_codec_driver hdmi_hda_codec = { .idle_bias_off = true, }; +static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + /* chmap is already set to 0 in caller */ + if (!pin) + return; + + memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap)); +} + +static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap, int prepared) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + mutex_lock(&pin->lock); + pin->chmap_set = true; + memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap)); + if (prepared) + hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid); + mutex_unlock(&pin->lock); +} + +static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + return pin ? true:false; +} + +static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + if (!pin && !pin->eld.eld_valid) + return 0; + + return pin->eld.info.spk_alloc; +} + static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) { struct hdac_device *codec = &edev->hdac; @@ -1538,6 +1634,10 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) edev->private_data = hdmi_priv; snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap); + hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; + hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; + hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; + hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; dev_set_drvdata(&codec->dev, edev); -- cgit v0.10.2 From 7e12dc87ac59963cf1765fb8272412db19004987 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:35 +0530 Subject: ASoC: Intel: Skylake: Add multichannel support for HDMI Channel max is changed to 8 from stereo to support multichannel capability for HDMI devices. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index dab0900..8de9212 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { .rate_min = 8000, .rate_max = 48000, .channels_min = 1, - .channels_max = HDA_QUAD, + .channels_max = 8, .buffer_bytes_max = AZX_MAX_BUF_SIZE, .period_bytes_min = 128, .period_bytes_max = AZX_MAX_BUF_SIZE / 2, @@ -682,7 +682,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI1 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -697,7 +697,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI2 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -712,7 +712,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI3 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -765,7 +765,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp1 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE, @@ -777,7 +777,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp2 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | @@ -790,7 +790,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp3 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | -- cgit v0.10.2 From ea5a137d0fe263854ae6267a0fa208c544d83452 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 14 Apr 2016 10:07:36 +0530 Subject: ASoC: Intel: Skylake: Update channel map based on runtime params Default channel map is set for 2 channels. Fix the channel map based on runtime params to support multichannel. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e7..8fceb7a 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx, dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg); } +static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs) +{ + int slot_map = 0xFFFFFFFF; + int start_slot = 0; + int i; + + for (i = 0; i < chs; i++) { + /* + * For 2 channels with starting slot as 0, slot map will + * look like 0xFFFFFF10. + */ + slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i))); + start_slot++; + } + fmt->ch_map = slot_map; +} + static void skl_tplg_update_params(struct skl_module_fmt *fmt, struct skl_pipe_params *params, int fixup) { if (fixup & SKL_RATE_FIXUP_MASK) fmt->s_freq = params->s_freq; - if (fixup & SKL_CH_FIXUP_MASK) + if (fixup & SKL_CH_FIXUP_MASK) { fmt->channels = params->ch; + skl_tplg_update_chmap(fmt, fmt->channels); + } if (fixup & SKL_FMT_FIXUP_MASK) { fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt); -- cgit v0.10.2 From 57dd5414a087991d427067b32dc3324af61b1c8b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Apr 2016 11:49:04 +0200 Subject: ALSA: usb-audio: Limit retrying sample rate reads There are many USB audio devices with buggy firmware that don't react with the sample rate reading properly. This often results in the flood of error messages and slowing down the operation. The sample rate read back is basically only for confirming the sample rate setup, and it's not critically important. As a compromise, in this patch, we stop the sample rate read back once when the device gives errors more than tolerance (twice, as of now). This should improve most of error cases while we still can catch the firmware bugginess. Signed-off-by: Takashi Iwai diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf..26dd5f2 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c..89b6853 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -50,6 +50,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ -- cgit v0.10.2 From 3cc6185bcccff32df41faa97d592a99d258db185 Mon Sep 17 00:00:00 2001 From: Caleb Crome Date: Mon, 25 Apr 2016 11:36:18 -0700 Subject: ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list The CCSR_SSI_SOR is a register that clears the TX and/or the RX fifo on the i.MX SSI port. The fsl_ssi_trigger writes this register in order to clear the fifo at trigger time. However, since the CCSR_SSI_SOR register is not in the volatile list, the caching mechanism prevented the register write in the trigger function. This caused the fifo to not be cleared (because the value was unchanged from the last time the register was written), and thus causes the channels in both TDM or simple I2S mode to slip and be in the wrong time slots on SSI restart. This has gone unnoticed for so long because with simple stereo mode, the consequence is that left and right are swapped, which isn't that noticeable. However, it's catestrophic in some systems that require the channels to be in the right slots. Signed-off-by: Caleb Crome Suggested-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de10..08dcbbf 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; -- cgit v0.10.2 From 823ecdd684e28d4e71686fc8787b6d31b1223382 Mon Sep 17 00:00:00 2001 From: Jim Lodes Date: Mon, 25 Apr 2016 11:08:10 -0500 Subject: ASoC: davinci-mcasp: Fix overwriting of ahclkx The mcasp davinci_mcasp_set_dai_fmt function was overriding ahclkx input/output status that had already been set by the davinci_mcasp_set_sysclk function. This commit removes clearing of the ahclkx input/output status from davinci_mcasp_set_dai_fmt. Signed-off-by: Jim Lodes Signed-off-by: J.D. Schroeder Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e132498..a1197ad 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: -- cgit v0.10.2 From 95536d8c29985167e745ff0d8c7cd7dcf4318e6b Mon Sep 17 00:00:00 2001 From: "Dharageswari.R" Date: Thu, 28 Apr 2016 18:45:25 +0530 Subject: ASoC: Intel: Skylake: Fix the NULL pointer exception in dsp_clean up If request firmware fails at init, the code loader DMA allocation can be NULL, so check for boot complete before freeing up these resources Signed-off-by: Dharageswari R Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index 2962ef2..13c1985 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -336,8 +336,6 @@ void skl_dsp_free(struct sst_dsp *dsp) skl_ipc_int_disable(dsp); free_irq(dsp->irq, dsp); - dsp->cl_dev.ops.cl_cleanup_controller(dsp); - skl_cldma_int_disable(dsp); skl_ipc_op_int_disable(dsp); skl_ipc_int_disable(dsp); diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index bec4a7c..13ec8d5 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -454,6 +454,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) skl_clear_module_table(ctx->dsp); skl_ipc_free(&ctx->ipc); ctx->dsp->ops->free(ctx->dsp); + if (ctx->boot_complete) { + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + skl_cldma_int_disable(ctx->dsp); + } } EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup); -- cgit v0.10.2 From 76222d6dd2e64c895735ab271ecc8b0df568981d Mon Sep 17 00:00:00 2001 From: Mousumi Jana Date: Thu, 28 Apr 2016 18:45:26 +0530 Subject: ASoC: Intel: Skylake: Fix memory leak during init instance param_data variable is allocated during set module format of init instance is not getting freed and hence can cause a memory leak. So free it up. Signed-off-by: Mousumi Jana Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index e3d149c..226db84 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -864,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx, return ret; } mconfig->m_state = SKL_MODULE_INIT_DONE; - + kfree(param_data); return ret; } -- cgit v0.10.2 From 1a13b1fafffd41c12a7068c4aa74f5a1d2210a07 Mon Sep 17 00:00:00 2001 From: "Dharageswari.R" Date: Thu, 28 Apr 2016 18:45:27 +0530 Subject: ASoC: Intel: Skylake: Prevent sending Set DMA Control IPC if the widget is "On" If widget of a playback or capture DAI is already On, then no need not send the Set DMA Control IPC message to firmware. Signed-off-by: Dharageswari R Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index dab0900..4fcf5f8 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream, struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - if ((dai->playback_active > 1) || (dai->capture_active > 1)) + if (dai->playback_widget->power || dai->capture_widget->power) return 0; mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream); -- cgit v0.10.2 From 9a655db0201ef523683d700cb3f4508c08bc9d8c Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 28 Apr 2016 18:45:28 +0530 Subject: ASoC: Intel: Skylake: Suspend PCMs when marked as active suspend For 'ignore_suspend' cases we need to keep DSP and pipes On, but can suspend the stream and pause the DMA as we are not rendering data during the suspended time. For this we can check the dai widget ignore_suspend flag in trigger suspend/resume, and start and stop the host DMA and host copier pipelines. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 4fcf5f8..b0e7797 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl_module_cfg *mconfig; struct hdac_ext_bus *ebus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct snd_soc_dapm_widget *w; int ret; mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); if (!mconfig) return -EIO; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - skl_pcm_prepare(substream, dai); - /* - * enable DMA Resume enable bit for the stream, set the dpib - * & lpib position to resune before starting the DMA - */ - snd_hdac_ext_stream_drsm_enable(ebus, true, - hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib); - snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + if (!w->ignore_suspend) { + skl_pcm_prepare(substream, dai); + /* + * enable DMA Resume enable bit for the stream, set the + * dpib & lpib position to resume before starting the + * DMA + */ + snd_hdac_ext_stream_drsm_enable(ebus, true, + hdac_stream(stream)->index); + snd_hdac_ext_stream_set_dpibr(ebus, stream, + stream->dpib); + snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + } case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return ret; ret = skl_decoupled_trigger(substream, cmd); - if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) { + if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ stream->dpib = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + -- cgit v0.10.2 From 551f4bc86807637098786c78afb78418ada4aa1f Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 28 Apr 2016 18:45:29 +0530 Subject: ASoC: Intel: Boards: remove ignore_suspend for WoV streams On WoV we can suspend the DMA and keep the DSP pipelines only On, so remove the ignore_suspend for WoV streams but keep them for WoV endpoints. This helps in achieving better power by suspending DMAs Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b7..ca8063d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -391,7 +391,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99..a0e3a3f 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -440,7 +440,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a..ef5b17f 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -317,7 +317,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, }, -- cgit v0.10.2 From 9ee78757d5dae51decc881b293a39a605c9a6df2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 2 May 2016 13:57:36 +0100 Subject: ASoC: wm_adsp: Add support for TLV based binary controls This patch adds support for the arbitrary length TLV based binary controls. This allows users to properly access controls that are more than 512 bytes in length. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3ac2e1f..f835277 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -435,6 +437,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +714,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +773,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +792,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +857,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +882,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +965,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -1032,11 +1125,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { -- cgit v0.10.2 From 8f658815da156a9239b98b34e5ba1d3db71a2f6e Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 3 May 2016 10:42:58 +0300 Subject: ASoC: hdac_hdmi: Potential NULL deref in hdac_hdmi_get_spk_alloc() We intended || here instead of &&. The original code potentially leads to a NULL dereference. Fixes: 2889099eb8cd ('ASoC: hdac_hdmi: Register chmap controls and ops') Signed-off-by: Dan Carpenter Reviewd-by: Takashi Sakamoto Acked-by: Vinod Koul Tested-by: Sachin Mokashi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 0ed3975..f1170e0 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1614,7 +1614,7 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_pin *pin = pcm->pin; - if (!pin && !pin->eld.eld_valid) + if (!pin || !pin->eld.eld_valid) return 0; return pin->eld.info.spk_alloc; -- cgit v0.10.2 From edd713509ae46ffcf178e3b1431af1ca202be8ba Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 17:11:55 +0100 Subject: ASoC: wm_adsp: Move compr_attach/attached functions Move wm_adsp_compr_attach and wm_adsp_compr_attached functions so they will stay logically grouped with similar functions after some additional changes. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 630ebcd..42fc469 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2452,6 +2452,25 @@ void wm_adsp2_remove(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_remove); +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + + return 0; +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2810,25 +2829,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } -static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) -{ - return compr->buf != NULL; -} - -static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) -{ - /* - * Note this will be more complex once each DSP can support multiple - * streams - */ - if (!compr->dsp->buffer) - return -EINVAL; - - compr->buf = compr->dsp->buffer; - - return 0; -} - int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) { struct wm_adsp_compr *compr = stream->runtime->private_data; -- cgit v0.10.2 From 721be3be2f75c69cf0f2d7826007a6eefee7dac3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 17:11:56 +0100 Subject: ASoC: wm_adsp: Detach compressed stream on free If someone powers down the DSP core (through routing changes say) whilst a compressed record is in progress we can end up using a freed pointer to the buffer object. When a compressed audio stream is triggered we attach it to a buffer on a physical DSP. This patch adds a detach of the buffer from the stream when the stream is freed or when the DSP is powered down which avoids the situation where we use a buffer when it is no longer valid. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 42fc469..a07bd7c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -273,8 +273,11 @@ struct wm_adsp_buffer { __be32 words_written[2]; /* total words written (64 bit) */ }; +struct wm_adsp_compr; + struct wm_adsp_compr_buf { struct wm_adsp *dsp; + struct wm_adsp_compr *compr; struct wm_adsp_buffer_region *regions; u32 host_buf_ptr; @@ -2467,10 +2470,26 @@ static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) return -EINVAL; compr->buf = compr->dsp->buffer; + compr->buf->compr = compr; return 0; } +static void wm_adsp_compr_detach(struct wm_adsp_compr *compr) +{ + if (!compr) + return; + + /* Wake the poll so it can see buffer is no longer attached */ + if (compr->stream) + snd_compr_fragment_elapsed(compr->stream); + + if (wm_adsp_compr_attached(compr)) { + compr->buf->compr = NULL; + compr->buf = NULL; + } +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2524,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream) mutex_lock(&dsp->pwr_lock); + wm_adsp_compr_detach(compr); dsp->compr = NULL; kfree(compr->raw_buf); @@ -2820,6 +2840,8 @@ err_buffer: static int wm_adsp_buffer_free(struct wm_adsp *dsp) { if (dsp->buffer) { + wm_adsp_compr_detach(dsp->buffer->compr); + kfree(dsp->buffer->regions); kfree(dsp->buffer); -- cgit v0.10.2 From a6e806c49e3265494ac6fe6ec88ed5c010652e0d Mon Sep 17 00:00:00 2001 From: John Keeping Date: Wed, 4 May 2016 17:21:56 +0100 Subject: ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables" This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e204..34743ec 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- cgit v0.10.2 From 7e885d211f023dfd201fad8246bbf3c3bd126c61 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Wed, 4 May 2016 17:21:57 +0100 Subject: ASoC: rockchip: Revert "ASoC: rockchip: i2s: separate capture and playback" This reverts commit eba65d179c1149cf79e68608d452631f33d7f017. This broke audio on Veyron Jerry Chromebooks and I now cannot reproduce the problem I was trying to fix even with this commit reverted, so it seems that this was completely the wrong thing to do. Reported-by: Enric Balletbo Serra Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 34743ec..574c6af 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); i2s->tx_start = true; } else { @@ -92,23 +92,27 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_STOP); - - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC, - I2S_CLR_TXC); + if (!i2s->rx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_TXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } @@ -124,8 +128,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); i2s->rx_start = true; } else { @@ -134,23 +138,27 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_STOP); - - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_RXC, - I2S_CLR_RXC); + if (!i2s->tx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_RXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } -- cgit v0.10.2 From 381437dd0bd590902320b97e6512792b075becd4 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 5 May 2016 11:13:31 +0800 Subject: ASoC: rt5645: polling jd status in all conditions We only polling jd status when rt5645->pdata.jd_invert is true. However, it should be done at all time since there will be no interrupt for jd if we press a headset button and remove the headset at the same time. Signed-off-by: Bard Liao Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dff706a..3c6594d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (btn_type == 0)/* button release */ report = rt5645->jack_type; else { - if (rt5645->pdata.jd_invert) { - mod_timer(&rt5645->btn_check_timer, - msecs_to_jiffies(100)); - } + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); } break; @@ -3816,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); } + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); -- cgit v0.10.2 From 5181365f5312d67dcdc9e4bc22516c48a83c8754 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 5 May 2016 11:53:06 +0100 Subject: ASoC: da7219: Add initial ACPI id for device This adds "DLGS7219" ACPI id for the codec. Signed-off-by: Adam Thomson Tested-by: Sathyanarayana Nujella Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index caea2ee..17e2119 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1426,6 +1426,12 @@ static const struct of_device_id da7219_of_match[] = { }; MODULE_DEVICE_TABLE(of, da7219_of_match); +static const struct acpi_device_id da7219_acpi_match[] = { + { .id = "DLGS7219", }, + { } +}; +MODULE_DEVICE_TABLE(acpi, da7219_acpi_match); + static enum da7219_micbias_voltage da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) { @@ -1955,6 +1961,7 @@ static struct i2c_driver da7219_i2c_driver = { .driver = { .name = "da7219", .of_match_table = of_match_ptr(da7219_of_match), + .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe = da7219_i2c_probe, .remove = da7219_i2c_remove, -- cgit v0.10.2 From 1593af62b694b3638edf577e3b763fa1a4ca3d76 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 May 2016 19:33:58 -0300 Subject: ASoC: fsl_sai: Introduce a compatible string for MX6UL MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..777b941 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,8 +7,8 @@ codec/DSP interfaces. Required properties: - - compatible : Compatible list, contains "fsl,vf610-sai" or - "fsl,imx6sx-sai". + - compatible : Compatible list, contains "fsl,vf610-sai", + "fsl,imx6sx-sai" or "fsl,imx6ul-sai" - reg : Offset and length of the register set for the device. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..d8b673f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- cgit v0.10.2 From 4d2458507d0b465c62ae80f3e81b8c008ec96b05 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 May 2016 19:33:59 -0300 Subject: ASoC: fsl_sai: Allow setting the SAI MCLK direction On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for configuring the direction of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce the "fsl,sai-mclk-direction-output" optional property to allow configuring the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 777b941..740b467 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,11 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - fsl,sai-mclk-direction-output: This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db..6835382 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,11 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) +#define IMX6UL_GPR1_SAI_MCLK_MASK (0x7 << 19) +#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \ + IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d8b673f..2147994 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include #include #include +#include +#include #include "fsl_sai.h" #include "imx-pcm.h" @@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + int index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -878,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + return PTR_ERR(gpr); + } + + index = of_alias_get_id(np, "sai"); + if (index < 0) + return index; + + regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index), + MCLK_DIR(index)); + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- cgit v0.10.2 From c286b3f9600b2ddc573208792d947e1a251c6b15 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 5 May 2016 11:19:19 +0530 Subject: ASoC: Intel: Skylake: Fix memory leak in nhlt init During skl_nhlt_init(), acpi obj pointer is allocated and never freed and remap address is not unmapped. To fix this we should release the ACPI obj and also unmap the nhlt address during cleanup of driver. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 14d1916e..7d73648 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void *skl_nhlt_init(struct device *dev) +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; struct nhlt_resource_desc *nhlt_ptr = NULL; + struct nhlt_acpi_table *nhlt_table = NULL; if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) { dev_err(dev, "Requested NHLT device not found\n"); @@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev) obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - - return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + nhlt_table = (struct nhlt_acpi_table *) + memremap(nhlt_ptr->min_addr, nhlt_ptr->length, MEMREMAP_WB); + ACPI_FREE(obj); + return nhlt_table; } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void *addr) +void skl_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap(addr); + memunmap((void *) nhlt); } static struct nhlt_specific_cfg *skl_get_specific_cfg( @@ -120,7 +123,7 @@ struct nhlt_specific_cfg struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_acpi_table *nhlt = skl->nhlt; u16 bps = (s_fmt == 16) ? 16 : 32; u8 j; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3982f55..83e985c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -643,7 +643,7 @@ static int skl_probe(struct pci_dev *pci, err = skl_machine_device_register(skl, (void *)pci_id->driver_data); if (err < 0) - goto out_free; + goto out_nhlt_free; err = skl_init_dsp(skl); if (err < 0) { @@ -693,6 +693,8 @@ out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); +out_nhlt_free: + skl_nhlt_free(skl->nhlt); out_free: skl->init_failed = 1; skl_free(ebus); @@ -743,6 +745,7 @@ static void skl_remove(struct pci_dev *pci) skl_free_dsp(skl); skl_machine_device_unregister(skl); skl_dmic_device_unregister(skl); + skl_nhlt_free(skl->nhlt); skl_free(ebus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 39e16fa..4b4b387 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -66,7 +66,7 @@ struct skl { struct platform_device *dmic_dev; struct platform_device *i2s_dev; - void *nhlt; /* nhlt ptr */ + struct nhlt_acpi_table *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ struct skl_dsp_resource resource; @@ -103,8 +103,8 @@ struct skl_dsp_ops { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void *addr); +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(struct nhlt_acpi_table *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); -- cgit v0.10.2 From b58cea7355875d6ae7aacb66c105f5c99f489909 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 May 2016 18:13:17 +0100 Subject: ASoC: da7129: Add missing include of acpi.h Reported-by: Stephen Rothwell Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 17e2119..5c93899 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -11,6 +11,7 @@ * option) any later version. */ +#include #include #include #include -- cgit v0.10.2 From f8ff65bce4fe9e94794beb21a3ba5e0cced43b1a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 30 Apr 2016 22:06:46 +0900 Subject: ALSA: dice: add support for M-Audio Profire 610 and perhaps 2626 M-Audio Profire 610 has an unexpected value in version field of its config ROM, thus ALSA dice driver is not assigned to the model due to a mismatch of modalias. This commit adds an entry to support the model. I expect the entry is also for Profire 2626. I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a part of Extended Application Protocol (EAP). $ cd linux-firewire-utils/src $ ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ------------------------------------------------------------ 400 04047689 bus_info_length 4, crc_length 4, crc 30345 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000d6c04 company_id 000d6c | 410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002 root directory ------------------------------------------------------------ 414 000695fe directory_length 6, crc 38398 418 03000d6c vendor 41c 8100000a --> descriptor leaf at 444 420 17000011 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ------------------------------------------------------------ 430 0004fb14 directory_length 4, crc 64276 434 12000d6c specifier id 438 130100d1 version 43c 17000011 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ------------------------------------------------------------ 444 0004b8e4 leaf_length 4, crc 47332 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4175 "M-Au" 454 64696f00 "dio" descriptor leaf at 458 ------------------------------------------------------------ 458 00053128 leaf_length 5, crc 12584 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 50726f46 "ProF" 468 69726520 "ire " 46c 36313000 "610" descriptor leaf at 470 ------------------------------------------------------------ 470 00053128 leaf_length 5, crc 12584 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 50726f46 "ProF" 480 69726520 "ire " 484 36313000 "610" $ cat /proc/asound/card1/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc0:000100000000 notification: 00000040 nick name: FW610 clock select: internal 48000 enable: 1 status: locked 48000 ext status: 00000040 sample rate: 48000 version: 1.0.4.0 clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\ ... Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96fe68f4..25e9f77 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -317,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); -- cgit v0.10.2 From 242658ff99ab9d87e704475ef78c3102ead344cf Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 2 May 2016 14:06:28 +0530 Subject: ALSA: compress: fix some typo Noticed two typos in Documentation, so fix them up Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 81476b4..8ba556a 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -184,7 +184,7 @@ Sequence flow for gapless would be: - Fill data of the first track - Trigger start - User-space finished sending all, -- Indicaite next track data by sending set_next_track +- Indicate next track data by sending set_next_track - Set metadata of the next track - then call partial_drain to flush most of buffer in DSP - Fill data of the next track -- cgit v0.10.2 From cec8f96e49d9be372fdb0c3836dcf31ec71e457e Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Tue, 3 May 2016 16:44:07 -0400 Subject: ALSA: timer: Fix leak in SNDRV_TIMER_IOCTL_PARAMS MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The stack object “tread” has a total size of 32 bytes. Its field “event” and “val” both contain 4 bytes padding. These 8 bytes padding bytes are sent to user without being initialized. Signed-off-by: Kangjie Lu Signed-off-by: Takashi Iwai diff --git a/sound/core/timer.c b/sound/core/timer.c index 0cfc028..306a93d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1737,6 +1737,7 @@ static int snd_timer_user_params(struct file *file, if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { if (tu->tread) { struct snd_timer_tread tread; + memset(&tread, 0, sizeof(tread)); tread.event = SNDRV_TIMER_EVENT_EARLY; tread.tstamp.tv_sec = 0; tread.tstamp.tv_nsec = 0; -- cgit v0.10.2 From 9a47e9cff994f37f7f0dbd9ae23740d0f64f9fe6 Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Tue, 3 May 2016 16:44:20 -0400 Subject: ALSA: timer: Fix leak in events via snd_timer_user_ccallback MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The stack object “r1” has a total size of 32 bytes. Its field “event” and “val” both contain 4 bytes padding. These 8 bytes padding bytes are sent to user without being initialized. Signed-off-by: Kangjie Lu Signed-off-by: Takashi Iwai diff --git a/sound/core/timer.c b/sound/core/timer.c index 306a93d..cc3c08d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1223,6 +1223,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, tu->tstamp = *tstamp; if ((tu->filter & (1 << event)) == 0 || !tu->tread) return; + memset(&r1, 0, sizeof(r1)); r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; -- cgit v0.10.2 From e4ec8cc8039a7063e24204299b462bd1383184a5 Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Tue, 3 May 2016 16:44:32 -0400 Subject: ALSA: timer: Fix leak in events via snd_timer_user_tinterrupt MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The stack object “r1” has a total size of 32 bytes. Its field “event” and “val” both contain 4 bytes padding. These 8 bytes padding bytes are sent to user without being initialized. Signed-off-by: Kangjie Lu Signed-off-by: Takashi Iwai diff --git a/sound/core/timer.c b/sound/core/timer.c index cc3c08d..e722022 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1266,6 +1266,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { + memset(&r1, 0, sizeof(r1)); r1.event = SNDRV_TIMER_EVENT_RESOLUTION; r1.tstamp = tstamp; r1.val = resolution; -- cgit v0.10.2 From 84d7a4470dbac0dd9389050100b54a1625d04264 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 4 May 2016 09:27:37 +0300 Subject: ALSA: isa/wavefront: prevent some out of bound writes "header->number" can be up to USHRT_MAX and it comes from the ioctl so it needs to be capped. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff..718d5e3 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics -- cgit v0.10.2 From dcd4f0db6141d6bf2cb897309d5d6f53d1b1696f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 4 May 2016 15:50:18 +0800 Subject: ALSA: hda/realtek - New codecs support for ALC234/ALC274/ALC294 Support new codecs for ALC234/ALC274/ALC294. This three codecs was the same IC. But bonding is not the same. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac4490a..a2fdf5f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -342,6 +342,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + alc_update_coef_idx(codec, 0x10, 1<<15, 0); + break; case 0x10ec0662: if ((coef & 0x00f0) == 0x0030) alc_update_coef_idx(codec, 0x4, 1<<10, 0); /* EAPD Ctrl */ @@ -2647,6 +2652,7 @@ enum { ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, ALC269_TYPE_ALC225, + ALC269_TYPE_ALC294, }; /* @@ -2677,6 +2683,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: case ALC269_TYPE_ALC225: + case ALC269_TYPE_ALC294: ssids = alc269_ssids; break; default: @@ -6028,6 +6035,11 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0225: spec->codec_variant = ALC269_TYPE_ALC225; break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + spec->codec_variant = ALC269_TYPE_ALC294; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -6929,6 +6941,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), @@ -6939,6 +6952,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269), HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269), HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0274, "ALC274", patch_alc269), HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269), HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269), HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269), @@ -6951,6 +6965,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269), HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), -- cgit v0.10.2 From 748a1ccc433f1c4edb214fff4b340af9c1da3f88 Mon Sep 17 00:00:00 2001 From: Oliver Neukum Date: Wed, 4 May 2016 14:18:39 +0200 Subject: ALSA: usb-audio: correct speed checking Allow handling SS+ USB devices correctly. Signed-off-by: Oliver Neukum Signed-off-by: Takashi Iwai diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bb..c4665dc 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -351,6 +351,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: break; default: dev_err(&dev->dev, "unknown device speed %d\n", snd_usb_get_speed(dev)); @@ -451,6 +452,9 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_SUPER: strlcat(card->longname, ", super speed", sizeof(card->longname)); break; + case USB_SPEED_SUPER_PLUS: + strlcat(card->longname, ", super speed plus", sizeof(card->longname)); + break; default: break; } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 51ed1ac..7712e2b 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -120,6 +120,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: if (get_endpoint(alts, 0)->bInterval >= 1 && get_endpoint(alts, 0)->bInterval <= 4) return get_endpoint(alts, 0)->bInterval - 1; -- cgit v0.10.2 From 89e448b33a101dfc7218f3181bc3095af850db7a Mon Sep 17 00:00:00 2001 From: Oliver Neukum Date: Wed, 4 May 2016 14:18:40 +0200 Subject: ALSA: usb-midi: correct speed checking Allow for SS+ USB devices Signed-off-by: Oliver Neukum Signed-off-by: Takashi Iwai diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 47de8af..7ba9292 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -911,6 +911,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, switch (snd_usb_get_speed(ep->umidi->dev)) { case USB_SPEED_HIGH: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: count = 1; break; default: -- cgit v0.10.2 From 94e9080ce22d9fb7c0a4361a5890b2c6affc9b1b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 5 May 2016 11:24:42 +0530 Subject: ALSA: hda: fix the missing ptr initialization ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c..64de0a3 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- cgit v0.10.2 From 38b19ed7f81ec930f3ad2066ae088f574970c814 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 5 May 2016 11:24:43 +0530 Subject: ALSA: hda: fix to wait for RIRB & CORB DMA to set If the DMAs are not being quiesced properly, it may lead to stability issues, so the recommendation is to wait till DMAs are stopped. After setting the stop bit of RIRB/CORB DMA, we should wait for stop bit to be set. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 8c48623..9fee464 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -80,6 +80,22 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io); +/* wait for cmd dmas till they are stopped */ +static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus) +{ + unsigned long timeout; + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN) + && time_before(jiffies, timeout)) + udelay(10); + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN) + && time_before(jiffies, timeout)) + udelay(10); +} + /** * snd_hdac_bus_stop_cmd_io - clean up CORB/RIRB buffers * @bus: HD-audio core bus @@ -90,6 +106,7 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus) /* disable ringbuffer DMAs */ snd_hdac_chip_writeb(bus, RIRBCTL, 0); snd_hdac_chip_writeb(bus, CORBCTL, 0); + hdac_wait_for_cmd_dmas(bus); /* disable unsolicited responses */ snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0); spin_unlock_irq(&bus->reg_lock); -- cgit v0.10.2 From 73fc7f080105b16d9f189c82d0092f22107cc67b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 21:12:44 +0900 Subject: ALSA: firewire-lib: compute the value of second field in cycle count for IT context In callback function of isochronous context, u32 variable is passed for cycle count. The value of this variable comes from DMA descriptors of 1394 Open Host Controller Interface (1394 OHCI). In the specification, DMA descriptors transport lower 3 bits for second field and full cycle field in 16 bits field, therefore 16 bits of the u32 variable are available. The value for second is modulo 8, and the value for cycle is modulo 8,000. Currently, ALSA firewire-lib module don't use the value of the second field, because the value is useless to calculate presentation timestamp in IEC 61883-6. However, the value may be useful for debugging. In later commit, it will be printed with the other parameters for debugging. This commit makes this module to handle the whole cycle count including second. The value is calculated by cycle unit. The existed code is already written with ignoring the value of second, thus this commit causes no issues. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4484242..46f1167 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -548,26 +548,44 @@ end: return 0; } -static void out_stream_callback(struct fw_iso_context *context, u32 cycle, +/* + * In CYCLE_TIMER register of IEEE 1394, 7 bits are used to represent second. On + * the other hand, in DMA descriptors of 1394 OHCI, 3 bits are used to represent + * it. Thus, via Linux firewire subsystem, we can get the 3 bits for second. + */ +static inline u32 compute_cycle_count(u32 tstamp) +{ + return (((tstamp >> 13) & 0x07) * 8000) + (tstamp & 0x1fff); +} + +static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) +{ + cycle += addend; + if (cycle >= 8 * CYCLES_PER_SECOND) + cycle -= 8 * CYCLES_PER_SECOND; + return cycle; +} + +static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; unsigned int i, syt, packets = header_length / 4; unsigned int data_blocks; + u32 cycle; if (s->packet_index < 0) return; - /* - * Compute the cycle of the last queued packet. - * (We need only the four lowest bits for the SYT, so we can ignore - * that bits 0-11 must wrap around at 3072.) - */ - cycle += QUEUE_LENGTH - packets; + cycle = compute_cycle_count(tstamp); + + /* Align to actual cycle count for the last packet. */ + cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); for (i = 0; i < packets; ++i) { - syt = calculate_syt(s, ++cycle); + cycle = increment_cycle_count(cycle, 1); + syt = calculate_syt(s, cycle); data_blocks = calculate_data_blocks(s, syt); if (handle_out_packet(s, data_blocks, syt) < 0) { @@ -580,7 +598,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 cycle, fw_iso_context_queue_flush(s->context); } -static void in_stream_callback(struct fw_iso_context *context, u32 cycle, +static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { @@ -650,7 +668,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, } /* processing is done by master callback */ -static void slave_stream_callback(struct fw_iso_context *context, u32 cycle, +static void slave_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { @@ -659,7 +677,7 @@ static void slave_stream_callback(struct fw_iso_context *context, u32 cycle, /* this is executed one time */ static void amdtp_stream_first_callback(struct fw_iso_context *context, - u32 cycle, size_t header_length, + u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; @@ -678,7 +696,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, else context->callback.sc = out_stream_callback; - context->callback.sc(context, cycle, header_length, header, s); + context->callback.sc(context, tstamp, header_length, header, s); } /** -- cgit v0.10.2 From f90e2dedf7f47ff4f2f757188a0360fbf526a81e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 21:12:45 +0900 Subject: ALSA: firewire-lib: compute the value of second field in cycle count for IR context In callback function of isochronous context, modules can queue packets to indicated isochronous cycles. Although the cycle to queue a packet is deterministic by calculation, this module doesn't implement the calculation because it's useless for processing. In future, the cycle count is going to be printed with the other parameters for debugging. This commit is the preparation. The cycle count is computed by cycle unit, and correctly arranged to corresponding packets. The calculated count is used in later commit. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 46f1167..4d86da0 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -566,6 +566,13 @@ static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) return cycle; } +static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) +{ + if (cycle < subtrahend) + cycle += 8 * CYCLES_PER_SECOND; + return cycle - subtrahend; +} + static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) @@ -607,6 +614,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, unsigned int payload_quadlets, max_payload_quadlets; unsigned int data_blocks; __be32 *buffer, *headers = header; + u32 cycle; if (s->packet_index < 0) return; @@ -614,10 +622,16 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, /* The number of packets in buffer */ packets = header_length / IN_PACKET_HEADER_SIZE; + cycle = compute_cycle_count(tstamp); + + /* Align to actual cycle count for the last packet. */ + cycle = decrement_cycle_count(cycle, packets); + /* For buffer-over-run prevention. */ max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4; for (p = 0; p < packets; p++) { + cycle = increment_cycle_count(cycle, 1); buffer = s->buffer.packets[s->packet_index].buffer; /* The number of quadlets in this packet */ -- cgit v0.10.2 From 0c95c1d6197f3edd3f6ef76f927d67e8ec0794ed Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 21:12:46 +0900 Subject: ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data When audio and music units have some quirks in their sequence of packet, it's really hard for non-owners to identify the quirks. Although developers need dumps for sequence of packets, it's difficult for users who have no knowledges and no equipments for this purpose. This commit adds tracepoints for this situation. When users encounter the issue, they can dump a part of packet data via Linux tracing framework as long as using drivers in ALSA firewire stack. Additionally, tracepoints for outgoing packets will be our help to check and debug packet processing of ALSA firewire stack. This commit newly adds 'snd_firewire_lib' subsystem with 'in_packet' and 'out_packet' events. In the events, some attributes of packets and the index of packet managed by this module are recorded per packet. This is an usage: $ trace-cmd record -e snd_firewire_lib:out_packet \ -e snd_firewire_lib:in_packet /sys/kernel/tracing/events/snd_firewire_lib/out_packet/filter /sys/kernel/tracing/events/snd_firewire_lib/in_packet/filter Hit Ctrl^C to stop recording ^C $ trace-cmd report trace.dat ... 23647.033934: in_packet: 01 4073 ffc0 ffc1 00 000f0040 9001b2d1 122 44 23647.033936: in_packet: 01 4074 ffc0 ffc1 00 000f0048 9001c83b 122 45 23647.033937: in_packet: 01 4075 ffc0 ffc1 00 000f0050 9001ffff 002 46 23647.033938: in_packet: 01 4076 ffc0 ffc1 00 000f0050 9001e1a6 122 47 23647.035426: out_packet: 01 4123 ffc1 ffc0 01 010f00d0 9001fb40 122 17 23647.035428: out_packet: 01 4124 ffc1 ffc0 01 010f00d8 9001ffff 002 18 23647.035429: out_packet: 01 4125 ffc1 ffc0 01 010f00d8 900114aa 122 19 23647.035430: out_packet: 01 4126 ffc1 ffc0 01 010f00e0 90012a15 122 20 (Here, some common fields are omitted so that a line to be within 80 characters.) ... One line represent one packet. The legend for the last nine fields is: - The second of cycle scheduled for the packet - The count of cycle scheduled for the packet - The ID of node as source (hex) - Some devices transfer packets with invalid source node ID in their CIP header. - The ID of node as destination (hex) - The value is not in CIP header of packets. - The value of isochronous channel - The first quadlet of CIP header (hex) - The second quadlet of CIP header (hex) - The number of included quadlets - The index of packet in a buffer maintained by this module This is an example to parse these lines from text file by Python3 script: \#!/usr/bin/env python3 import sys def parse_ts(second, cycle, syt): offset = syt & 0xfff syt >>= 12 if cycle & 0x0f > syt: cycle += 0x10 cycle &= 0x1ff0 cycle |= syt second += cycle // 8000 cycle %= 8000 # In CYCLE_TIMER of 1394 OHCI, second is represented in 8 bit. second %= 128 return (second, cycle, offset) def calc_ts(second, cycle, offset): ts = offset ts += cycle * 3072 # In DMA descriptor of 1394 OHCI, second is represented in 3 bit. ts += (second % 8) * 8000 * 3072 return ts def subtract_ts(minuend, subtrahend): # In DMA descriptor of 1394 OHCI, second is represented in 3 bit. if minuend < subtrahend: minuend += 8 * 8000 * 3072 return minuend - subtrahend if len(sys.argv) != 2: print('At least, one argument is required for packet dump.') sys.exit() filename = sys.argv[1] data = [] prev = 0 with open(filename, 'r') as f: for line in f: pos = line.find('packet:') if pos < 0: continue pos += len('packet:') line = line[pos:].strip() fields = line.split(' ') datum = [] datum.append(fields[8]) syt = int(fields[6][4:], 16) # Empty packet in IEC 61883-1, or NODATA in IEC 61883-6 if syt == 0xffff: data_blocks = 0 else: payload_size = int(fields[7], 10) data_block_size = int(fields[5][2:4], 16) data_blocks = (payload_size - 2) / data_block_size datum.append(data_blocks) second = int(fields[0], 10) cycle = int(fields[1], 10) start = (second << 25) | (cycle << 12) datum.append('0x{0:08x}'.format(start)) start = calc_ts(second, cycle, 0) datum.append("0x" + fields[5]) datum.append("0x" + fields[6]) if syt == 0xffff: second = 0 cycle = 0 tick = 0 else: second, cycle, tick = parse_ts(second, cycle, syt) ts = calc_ts(second, cycle, tick) datum.append(start) datum.append(ts) if ts == 0: datum.append(0) datum.append(0) else: # Usual case, or a case over 8 seconds. if ts > start or start > 7 * 8000 * 3072: datum.append(subtract_ts(ts, start)) if ts > prev or start > 7 * 8000 * 3072: gap = subtract_ts(ts, prev) datum.append(gap) else: datum.append('backward') else: datum.append('invalid') prev = ts data.append(datum) sys.exit() The data variable includes array with these elements: - The index of the packet - The number of data blocks in the packet - The value of cycle count (hex) - The value of CIP header 1 (hex) - The value of CIP header 2 (hex) - The value of cycle count (tick) - The value of calculated presentation timestamp (tick) - The offset between the cycle count and presentation timestamp - The elapsed ticks from the previous presentation timestamp Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 003c090..0ee1fb1 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 0000000..425d1d7 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,98 @@ +/* + * amdtp-stream-trace.h - tracepoint definitions to dump a part of packet data + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include + +TRACE_EVENT(in_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets), + TP_ARGS(s, cycles, cip_header, payload_quadlets), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->node_id; + __entry->dest = fw_parent_device(s->unit)->card->node_id; + __entry->cip_header0 = cip_header[0]; + __entry->cip_header1 = cip_header[1]; + __entry->payload_quadlets = payload_quadlets; + __entry->index = s->packet_index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length), + TP_ARGS(s, cycles, cip_header, payload_length), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->card->node_id; + __entry->dest = fw_parent_device(s->unit)->node_id; + __entry->cip_header0 = be32_to_cpu(cip_header[0]); + __entry->cip_header1 = be32_to_cpu(cip_header[1]); + __entry->payload_quadlets = payload_length / 4; + __entry->index = s->packet_index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4d86da0..a22e559 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -19,6 +19,10 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* Always support Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ @@ -409,7 +413,7 @@ static inline int queue_in_packet(struct amdtp_stream *s) } static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, - unsigned int syt) + unsigned int cycle, unsigned int syt) { __be32 *buffer; unsigned int payload_length; @@ -428,8 +432,10 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, (syt & CIP_SYT_MASK)); s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; - payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; + + trace_out_packet(s, cycle, buffer, payload_length); + if (queue_out_packet(s, payload_length, false) < 0) return -EIO; @@ -443,7 +449,8 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, static int handle_in_packet(struct amdtp_stream *s, unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks, unsigned int syt) + unsigned int *data_blocks, unsigned int cycle, + unsigned int syt) { u32 cip_header[2]; unsigned int fmt, fdf; @@ -455,6 +462,8 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(s, cycle, cip_header, payload_quadlets); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -595,7 +604,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, syt = calculate_syt(s, cycle); data_blocks = calculate_data_blocks(s, syt); - if (handle_out_packet(s, data_blocks, syt) < 0) { + if (handle_out_packet(s, data_blocks, cycle, syt) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -647,7 +656,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, syt) < 0) { + &data_blocks, cycle, syt) < 0) { s->packet_index = -1; break; } @@ -655,7 +664,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, /* Process sync slave stream */ if (s->sync_slave && s->sync_slave->callbacked) { if (handle_out_packet(s->sync_slave, - data_blocks, syt) < 0) { + data_blocks, cycle, syt) < 0) { s->packet_index = -1; break; } -- cgit v0.10.2 From aeea2fdd9b623fcd6991ac3617ef6a3b646c2899 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 7 May 2016 20:04:52 -0300 Subject: MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers I would like to help reviewing FSL/NXP SoC sound drivers. Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown diff --git a/MAINTAINERS b/MAINTAINERS index 03e00c7..734bd0d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4661,6 +4661,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi M: Nicolin Chen M: Xiubo Li +R: Fabio Estevam L: alsa-devel@alsa-project.org (moderated for non-subscribers) L: linuxppc-dev@lists.ozlabs.org S: Maintained -- cgit v0.10.2 From 19357366633cfc53532b587180af3655f0e453f3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:39:14 +0300 Subject: ASoC: davinci-mcasp: Do not allow multiple streams in one direction Make sure that the user can not start multiple streams with the same direction. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e132498..020d866 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1230,11 +1230,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; -- cgit v0.10.2 From e099aeea639ce491d3cd1c3802fe34d98045ffd8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 14:59:07 +0100 Subject: ALSA: pcm: Fix poll error return codes We can't return a negative error code from the poll callback the return type is unsigned and is checked against the poll specific flags we need to return POLLERR if we encounter an error. Signed-off-by: Charles Keepax Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9106d8e..c61fd50 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLOUT | POLLWRNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); @@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLIN | POLLRDNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); -- cgit v0.10.2 From 0b92b0cdbe419575b2233c08192b2ad28e7dbcfa Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 14:59:08 +0100 Subject: ALSA: compress: Use snd_compr_get_poll on error path We have a function that returns the appropriate flags for the stream direction, so we should use it. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a9933c0..5268546 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -421,10 +421,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) retval = snd_compr_get_poll(stream); break; default: - if (stream->direction == SND_COMPRESS_PLAYBACK) - retval = POLLOUT | POLLWRNORM | POLLERR; - else - retval = POLLIN | POLLRDNORM | POLLERR; + retval = snd_compr_get_poll(stream) | POLLERR; break; } out: -- cgit v0.10.2 From 5bd05390ff084d7a1ea7efa8f8dc111c24b2454c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 14:59:09 +0100 Subject: ALSA: compress: Remove pointless NULL check stream can't be NULL here as we have just taken the address of it, so no need for the check. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 5268546..5215df2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -392,9 +392,8 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) if (snd_BUG_ON(!data)) return -EFAULT; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; mutex_lock(&stream->device->lock); if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { @@ -799,9 +798,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!data)) return -EFAULT; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; + mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): -- cgit v0.10.2 From 1d03f2bd56f3a45123a7572fb536c063068cfb83 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 14:59:10 +0100 Subject: ALSA: compress: Fix poll error return codes We can't return a negative error code from the poll callback the return type is unsigned and is checked against the poll specific flags we need to return POLLERR if we encounter an error. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 5215df2..f56f4e3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -391,13 +391,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) int retval = 0; if (snd_BUG_ON(!data)) - return -EFAULT; + return POLLERR; stream = &data->stream; mutex_lock(&stream->device->lock); if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { - retval = -EBADFD; + retval = snd_compr_get_poll(stream) | POLLERR; goto out; } poll_wait(f, &stream->runtime->sleep, wait); -- cgit v0.10.2 From 875f6fffa2e09b48fa07ecbf2e28dd2425b5ce01 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 4 May 2016 14:59:11 +0100 Subject: ALSA: compress: Replace complex if statement with switch A switch statement looks a bit cleaner than an if statement spread over 3 lines, as such update this to a switch. Signed-off-by: Charles Keepax Acked-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index f56f4e3..9b3334b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); /* write is allowed when stream is running or has been steup */ - if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_PREPARED && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_RUNNING: + break; + default: mutex_unlock(&stream->device->lock); return -EBADFD; } -- cgit v0.10.2 From 39f0ccde3624e7cf882faccf7f72a47b7a763bfb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 May 2016 17:47:37 +0200 Subject: ALSA: hda - Clarify CONFIG_SND_HDA_RECONFIG usages Since the recent rewrite of HD-audio infrastructure, CONFIG_SND_HDA_RECONFIG has a slightly different meaning. In the earlier versions, it implicitly assumed only the usage via hwdep sysfs entries. Meanwhile, in the recent version, this option is meant to enable the reconfig code in HD-audio driver, which may be used by the patch loader without hwdep interface. This patch tries to clarify the usage pattern a bit better, hopefully avoid the further confusion. Reported-by: Jochen Henneberg Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb02c2d..7f3b5ed 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -50,9 +50,13 @@ config SND_HDA_RECONFIG bool "Allow dynamic codec reconfiguration" help Say Y here to enable the HD-audio codec re-configuration feature. - This adds the sysfs interfaces to allow user to clear the whole - codec configuration, change the codec setup, add extra verbs, - and re-configure the codec dynamically. + It allows user to clear the whole codec configuration, change the + codec setup, add extra verbs, and re-configure the codec dynamically. + + Note that this item alone doesn't provide the sysfs interface, but + enables the feature just for the patch loader below. + If you need the traditional sysfs entries for the manual interaction, + turn on CONFIG_SND_HDA_HWDEP as well. config SND_HDA_INPUT_BEEP bool "Support digital beep via input layer" -- cgit v0.10.2 From 20d4b10730183a02851580f072bd9b0122873dc5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:42:29 +0300 Subject: ASoC: davinci-mcasp: Use defines for clkdiv IDs Instead of hardwired IDs add defines for the available dividers. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 020d866..adf1c39 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -547,14 +547,14 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, pm_runtime_get_sync(mcasp->dev); switch (div_id) { - case 0: /* MCLK divider */ + case MCASP_CLKDIV_AUXCLK: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; - case 1: /* BCLK divider */ + case MCASP_CLKDIV_BCLK: /* BCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, @@ -563,7 +563,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* + case MCASP_CLKDIV_BCLK_FS_RATIO: + /* * BCLK/LRCLK ratio descries how many bit-clock cycles * fit into one frame. The clock ratio is given for a * full period of data (for I2S format both left and diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a3be108..1e8787f 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -306,4 +306,9 @@ #define NUMEVT(x) (((x) & 0xFF) << 8) #define NUMDMA_MASK (0xFF) +/* clock divider IDs */ +#define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */ +#define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */ +#define MCASP_CLKDIV_BCLK_FS_RATIO 2 /* to set BCLK FS ration */ + #endif /* DAVINCI_MCASP_H */ -- cgit v0.10.2 From 226e73e23b6b7f7d6df47562a7555ddb121163cf Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:42:30 +0300 Subject: ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter Change the first parameter to struct davinci_mcasp* from struct snd_soc_dai* The function internally does not use or need the DAI information. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index adf1c39..99061c4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -540,11 +540,9 @@ out: return ret; } -static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, +static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, int div, bool explicit) { - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - pm_runtime_get_sync(mcasp->dev); switch (div_id) { case MCASP_CLKDIV_AUXCLK: /* MCLK divider */ @@ -592,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + + return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1); } static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -1056,7 +1056,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", ppm); - __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); + __davinci_mcasp_set_clkdiv(mcasp, 1, div, 0); } ret = mcasp_common_hw_param(mcasp, substream->stream, -- cgit v0.10.2 From 3e9bee11d83190b852d428b3e35a942c6e2293cd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:42:31 +0300 Subject: ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() Change the return value to error_pmm instead of the BCLK div and handle the divider configuration to McASP within the function when the set flag is true. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 99061c4..58fe112 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1000,9 +1000,9 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, } static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, - unsigned int bclk_freq, - int *error_ppm) + unsigned int bclk_freq, bool set) { + int error_ppm; int div = mcasp->sysclk_freq / bclk_freq; int rem = mcasp->sysclk_freq % bclk_freq; @@ -1014,13 +1014,18 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, rem = rem - bclk_freq; } } - if (error_ppm) - *error_ppm = - (div*1000000 + (int)div64_long(1000000LL*rem, - (int)bclk_freq)) - /div - 1000000; + error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) / div - 1000000; - return div; + if (set) { + if (error_ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + error_ppm); + + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0); + } + + return error_ppm; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -1045,18 +1050,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); - int ppm, div; if (mcasp->slot_width) sbits = mcasp->slot_width; - div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots, - &ppm); - if (ppm) - dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", - ppm); - - __davinci_mcasp_set_clkdiv(mcasp, 1, div, 0); + davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1167,7 +1165,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, davinci_mcasp_dai_rates[i]; int ppm; - davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { if (range.empty) { range.min = davinci_mcasp_dai_rates[i]; @@ -1206,8 +1205,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, if (rd->mcasp->slot_width) sbits = rd->mcasp->slot_width; - davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate, - &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, + sbits * slots * rate, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { snd_mask_set(&nfmt, i); count++; -- cgit v0.10.2 From ddecd1492de476488a92493510fb86c6ffe9acbd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:42:32 +0300 Subject: ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks If the McASP is used as clock master and the reference clock is AUXCLK we can have additional level of divider. The BCLK divider is limited to maximum 32, if the desired bclk can not be reached with this, the AUXCLK divider also needs to be used. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 58fe112..f390bb4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1003,13 +1003,31 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, unsigned int bclk_freq, bool set) { int error_ppm; - int div = mcasp->sysclk_freq / bclk_freq; - int rem = mcasp->sysclk_freq % bclk_freq; + unsigned int sysclk_freq = mcasp->sysclk_freq; + u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG); + int div = sysclk_freq / bclk_freq; + int rem = sysclk_freq % bclk_freq; + int aux_div = 1; + + if (div > (ACLKXDIV_MASK + 1)) { + if (reg & AHCLKXE) { + aux_div = div / (ACLKXDIV_MASK + 1); + if (div % (ACLKXDIV_MASK + 1)) + aux_div++; + + sysclk_freq /= aux_div; + div = sysclk_freq / bclk_freq; + rem = sysclk_freq % bclk_freq; + } else if (set) { + dev_warn(mcasp->dev, "Too fast reference clock (%u)\n", + sysclk_freq); + } + } if (rem != 0) { if (div == 0 || - ((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + ((sysclk_freq / div) - bclk_freq) > + (bclk_freq - (sysclk_freq / (div+1)))) { div++; rem = rem - bclk_freq; } @@ -1023,6 +1041,9 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, error_ppm); __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0); + if (reg & AHCLKXE) + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK, + aux_div, 0); } return error_ppm; -- cgit v0.10.2 From c71283cb682c28085125bea35e4c6149b538f5db Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:50 +0900 Subject: ALSA: bebob: drop reuse of incoming packet parameter for outgoing packet parameter Windows driver for BeBoB-based models mostly wait for transmitted packets, then transfer packets to the models. This looks for the relationship between incoming packets and outgoing packets to synchronize the sequence of presentation timestamp. However, the sequence between packets of both direction has no relationship. Even if receiving NO-DATA packets, the drivers transfer packets with meaningful value in SYT field. Additionally, the order of starting packets is always the same, independently of the source of clock. The corresponding driver is expected as a generator of presentation timestamp and these models can select it as a source of sampling clock. This commit drops reusing SYT sequence from ALSA bebob driver. The driver always transfer packets with presentation timestamp generated by ALSA firewire stack, without re-using the sequence of value in SYT field in incoming packets to outgoing packets. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 2a442a7..e7f1bb9 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -94,7 +94,6 @@ struct snd_bebob { bool connected; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 77cbb02..0141813 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,30 +484,6 @@ destroy_both_connections(struct snd_bebob *bebob) } static int -get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode) -{ - enum snd_bebob_clock_type src; - int err; - - err = snd_bebob_stream_get_clock_src(bebob, &src); - if (err < 0) - return err; - - switch (src) { - case SND_BEBOB_CLOCK_TYPE_INTERNAL: - case SND_BEBOB_CLOCK_TYPE_EXTERNAL: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - default: - case SND_BEBOB_CLOCK_TYPE_SYT: - *sync_mode = 0; - break; - } - - return 0; -} - -static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate) { @@ -584,8 +560,6 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; - struct amdtp_stream *master, *slave; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -593,22 +567,11 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (bebob->substreams_counter == 0) goto end; - err = get_sync_mode(bebob, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &bebob->tx_stream; - slave = &bebob->rx_stream; - } else { - master = &bebob->rx_stream; - slave = &bebob->tx_stream; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(bebob, master); + err = check_connection_used_by_others(bebob, &bebob->rx_stream); if (err < 0) goto end; @@ -618,11 +581,12 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) * At bus reset, connections should not be broken here. So streams need * to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag. */ - if (amdtp_streaming_error(master)) - amdtp_stream_stop(master); - if (amdtp_streaming_error(slave)) - amdtp_stream_stop(slave); - if (!amdtp_stream_running(master) && !amdtp_stream_running(slave)) + if (amdtp_streaming_error(&bebob->rx_stream)) + amdtp_stream_stop(&bebob->rx_stream); + if (amdtp_streaming_error(&bebob->tx_stream)) + amdtp_stream_stop(&bebob->tx_stream); + if (!amdtp_stream_running(&bebob->rx_stream) && + !amdtp_stream_running(&bebob->tx_stream)) break_both_connections(bebob); /* stop streams if rate is different */ @@ -635,16 +599,13 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - amdtp_stream_stop(master); - amdtp_stream_stop(slave); + amdtp_stream_stop(&bebob->rx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - bebob->master = master; - + if (!amdtp_stream_running(&bebob->rx_stream)) { /* * NOTE: * If establishing connections at first, Yamaha GO46 @@ -666,7 +627,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (err < 0) goto end; - err = start_stream(bebob, master, rate); + err = start_stream(bebob, &bebob->rx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP master stream:%d\n", err); @@ -685,15 +646,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) dev_err(&bebob->unit->device, "fail to ensure sampling rate: %d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } } /* wait first callback */ - if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->rx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; goto end; @@ -701,20 +663,21 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) } /* start slave if needed */ - if (!amdtp_stream_running(slave)) { - err = start_stream(bebob, slave, rate); + if (!amdtp_stream_running(&bebob->tx_stream)) { + err = start_stream(bebob, &bebob->tx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP slave stream:%d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } /* wait first callback */ - if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(slave); - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->tx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->tx_stream); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; } @@ -725,22 +688,12 @@ end: void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { - struct amdtp_stream *master, *slave; - - if (bebob->master == &bebob->rx_stream) { - slave = &bebob->tx_stream; - master = &bebob->rx_stream; - } else { - slave = &bebob->rx_stream; - master = &bebob->tx_stream; - } - if (bebob->substreams_counter == 0) { - amdtp_stream_pcm_abort(master); - amdtp_stream_stop(master); + amdtp_stream_pcm_abort(&bebob->rx_stream); + amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_pcm_abort(slave); - amdtp_stream_stop(slave); + amdtp_stream_pcm_abort(&bebob->tx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } -- cgit v0.10.2 From eb4a378fc99d876e98e01d67701c49343fae3e39 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:51 +0900 Subject: ALSA: fireworks: drop reuse of incoming packet parameter for ougoing packet parameter On Fireworks board module of Echo Audio, TSB43Cx43A (IceLynx Micro, iCEM) is used to process payload of isochronous packets. There's an public document of this chip[1]. This document is for firmware programmers to transfer/receive AMDTP with IEC60958 data format, however in clause 2.5, 2.6 and 2.7, we can see system design to utilize the sequence of value in SYT field of CIP header. In clause 2.3, we can see the specification of Audio Master Clock (MCLK) from iCEM. Well, this clock is actually not used for sampling clock. This can be confirmed when corresponding driver transfer random value as the sequence of SYT field. Even if in this case, the unit generates proper sound. Additionally, in unique command set for this board module, the format of CIP is changed; for IEC 61883-6 mode which we use, and for Windows Operating System. In the latter mode, the whole 32 bit field in second CIP header from Windows driver is used to represent counter of packets (NO-DATA code is still used for packets without data blocks). If the master clock was physically used by DSP on the board module, the Windows driver must have transferred correct sequence of SYT field. Furthermore, as long as seeing capacities of AudioFire2, AudioFire4, AudioFire8, AudioFirePre8 and AudioFire12, these models don't support SYT-Match clock source. Summary, we have no need to relate incoming/outgoing packets. This commit drops reusing SYT sequence of incoming packets for outgoing packets. [1] Using TSB43Cx43A: S/PDIF over 1394 (2003, Texus Instruments Incorporated) http://www.ti.com/analog/docs/litabsmultiplefilelist.tsp?literatureNumber=slla148&docCategoryId=1&familyId=361 Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 471c772..03ed352 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -84,7 +84,6 @@ struct snd_efw { unsigned int pcm_capture_channels[SND_EFW_MULTIPLIER_MODES]; unsigned int pcm_playback_channels[SND_EFW_MULTIPLIER_MODES]; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 425db8d..ee47924 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -121,23 +121,6 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) } static int -get_sync_mode(struct snd_efw *efw, enum cip_flags *sync_mode) -{ - enum snd_efw_clock_source clock_source; - int err; - - err = snd_efw_command_get_clock_source(efw, &clock_source); - if (err < 0) - return err; - - if (clock_source == SND_EFW_CLOCK_SOURCE_SYTMATCH) - return -ENOSYS; - - *sync_mode = CIP_SYNC_TO_DEVICE; - return 0; -} - -static int check_connection_used_by_others(struct snd_efw *efw, struct amdtp_stream *s) { struct cmp_connection *conn; @@ -208,9 +191,6 @@ end: int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) { - struct amdtp_stream *master, *slave; - unsigned int slave_substreams; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -218,32 +198,19 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (efw->playback_substreams == 0 && efw->capture_substreams == 0) goto end; - err = get_sync_mode(efw, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &efw->tx_stream; - slave = &efw->rx_stream; - slave_substreams = efw->playback_substreams; - } else { - master = &efw->rx_stream; - slave = &efw->tx_stream; - slave_substreams = efw->capture_substreams; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(efw, master); + err = check_connection_used_by_others(efw, &efw->rx_stream); if (err < 0) goto end; /* packet queueing error */ - if (amdtp_streaming_error(slave)) - stop_stream(efw, slave); - if (amdtp_streaming_error(master)) - stop_stream(efw, master); + if (amdtp_streaming_error(&efw->tx_stream)) + stop_stream(efw, &efw->tx_stream); + if (amdtp_streaming_error(&efw->rx_stream)) + stop_stream(efw, &efw->rx_stream); /* stop streams if rate is different */ err = snd_efw_command_get_sampling_rate(efw, &curr_rate); @@ -252,20 +219,17 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - stop_stream(efw, slave); - stop_stream(efw, master); + stop_stream(efw, &efw->tx_stream); + stop_stream(efw, &efw->rx_stream); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - efw->master = master; - + if (!amdtp_stream_running(&efw->rx_stream)) { err = snd_efw_command_set_sampling_rate(efw, rate); if (err < 0) goto end; - err = start_stream(efw, master, rate); + err = start_stream(efw, &efw->rx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP master stream:%d\n", err); @@ -274,12 +238,13 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) } /* start slave if needed */ - if (slave_substreams > 0 && !amdtp_stream_running(slave)) { - err = start_stream(efw, slave, rate); + if (efw->capture_substreams > 0 && + !amdtp_stream_running(&efw->tx_stream)) { + err = start_stream(efw, &efw->tx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP slave stream:%d\n", err); - stop_stream(efw, master); + stop_stream(efw, &efw->rx_stream); } } end: @@ -288,26 +253,11 @@ end: void snd_efw_stream_stop_duplex(struct snd_efw *efw) { - struct amdtp_stream *master, *slave; - unsigned int master_substreams, slave_substreams; - - if (efw->master == &efw->rx_stream) { - slave = &efw->tx_stream; - master = &efw->rx_stream; - slave_substreams = efw->capture_substreams; - master_substreams = efw->playback_substreams; - } else { - slave = &efw->rx_stream; - master = &efw->tx_stream; - slave_substreams = efw->playback_substreams; - master_substreams = efw->capture_substreams; - } - - if (slave_substreams == 0) { - stop_stream(efw, slave); + if (efw->capture_substreams == 0) { + stop_stream(efw, &efw->tx_stream); - if (master_substreams == 0) - stop_stream(efw, master); + if (efw->playback_substreams == 0) + stop_stream(efw, &efw->rx_stream); } } -- cgit v0.10.2 From 28e64f5176387bf5b9458d213650b90fa719be88 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:52 +0900 Subject: ALSA: firewire-tascam: drop reuse of incoming packet parameter for outgoing packet parameter In packet streaming protocol applied to TASCAM FireWire series, the value of SYT field in CIP header is always zero, therefore it has no meaning. There's no need to synchronize packets in both direction for the series. In current implementation of ALSA firewire stack, driver for the series uses incoming packet parameter for outgoing packet parameter to calculate the number of data blocks. This can be simplified because the task of corresponding driver is to transfer data blocks enough to sampling transfer frequency. This commit purges support of full duplex synchronization to prevent over-engineering implementation. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 0e6dd5c6..4ad3bd7 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -381,19 +381,17 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) return err; if (curr_rate != rate || - amdtp_streaming_error(&tscm->tx_stream) || - amdtp_streaming_error(&tscm->rx_stream)) { + amdtp_streaming_error(&tscm->rx_stream) || + amdtp_streaming_error(&tscm->tx_stream)) { finish_session(tscm); - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); release_resources(tscm); } - if (!amdtp_stream_running(&tscm->tx_stream)) { - amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE, - &tscm->tx_stream, &tscm->rx_stream); + if (!amdtp_stream_running(&tscm->rx_stream)) { err = keep_resources(tscm, rate); if (err < 0) goto error; @@ -406,27 +404,27 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_stream_start(&tscm->tx_stream, - tscm->tx_resources.channel, + err = amdtp_stream_start(&tscm->rx_stream, + tscm->rx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->tx_stream, + if (!amdtp_stream_wait_callback(&tscm->rx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } } - if (!amdtp_stream_running(&tscm->rx_stream)) { - err = amdtp_stream_start(&tscm->rx_stream, - tscm->rx_resources.channel, + if (!amdtp_stream_running(&tscm->tx_stream)) { + err = amdtp_stream_start(&tscm->tx_stream, + tscm->tx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->rx_stream, + if (!amdtp_stream_wait_callback(&tscm->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -435,8 +433,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; error: - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); finish_session(tscm); release_resources(tscm); -- cgit v0.10.2 From dec63cc8b65b04c0ebb5d82b6b399665d6d44dca Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:53 +0900 Subject: ALSA: firewire-lib: handle IT/IR contexts in each software interrupt context In clause 6.3 of IEC 61883-6:2000, there's an explanation about processing of presentation timestamp. In the clause, we can see "If a function block receives a CIP, processes it and subsequently re-transmits it, then the SYT of the outgoing CIP shall be the sum of the incoming SYT and the processing delay." ALSA firewire stack has an implementation to partly satisfy this specification. Developers assumed the stack to perform as an Audio function block[1]. Following to the assumption, current implementation of ALSA firewire stack use one software interrupt context to handle both of in/out packets. In most case, this is processed in 1394 OHCI IR context independently of the opposite context. Thus, this implementation uses longer CPU time in the software interrupt context. This is not better for whole system. Against the assumption, I confirmed that each ASIC for IEC 61883-1/6 doesn't necessarily expect it to the stack. Thus, current implementation of ALSA firewire stack includes over-engineering. This commit purges the implementation. As a result, packets of one direction are handled in one software interrupt context and spends minimum CPU time. [1] [alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062660.html Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index a22e559..3468419 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -91,7 +91,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, init_waitqueue_head(&s->callback_wait); s->callbacked = false; - s->sync_slave = NULL; s->fmt = fmt; s->process_data_blocks = process_data_blocks; @@ -650,54 +649,25 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, dev_err(&s->unit->device, "Detect jumbo payload: %02x %02x\n", payload_quadlets, max_payload_quadlets); - s->packet_index = -1; break; } syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, cycle, syt) < 0) { - s->packet_index = -1; + &data_blocks, cycle, syt) < 0) break; - } - - /* Process sync slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) { - if (handle_out_packet(s->sync_slave, - data_blocks, cycle, syt) < 0) { - s->packet_index = -1; - break; - } - } } - /* Queueing error or detecting discontinuity */ - if (s->packet_index < 0) { + /* Queueing error or detecting invalid payload. */ + if (p < packets) { + s->packet_index = -1; amdtp_stream_pcm_abort(s); - - /* Abort sync slave. */ - if (s->sync_slave) { - s->sync_slave->packet_index = -1; - amdtp_stream_pcm_abort(s->sync_slave); - } return; } - /* when sync to device, flush the packets for slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) - fw_iso_context_queue_flush(s->sync_slave->context); - fw_iso_context_queue_flush(s->context); } -/* processing is done by master callback */ -static void slave_stream_callback(struct fw_iso_context *context, u32 tstamp, - size_t header_length, void *header, - void *private_data) -{ - return; -} - /* this is executed one time */ static void amdtp_stream_first_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, @@ -714,8 +684,6 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, if (s->direction == AMDTP_IN_STREAM) context->callback.sc = in_stream_callback; - else if (s->flags & CIP_SYNC_TO_DEVICE) - context->callback.sc = slave_stream_callback; else context->callback.sc = out_stream_callback; diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704..da028b0 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -17,8 +17,6 @@ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or * SYT_INTERVAL samples, with these two types alternating so that * the overall sample rate comes out right. - * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is - * generated by in packets. Defaultly this driver generates timestamp. * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0. * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet * corresponds to the end of event in the packet. Out of IEC 61883. @@ -37,14 +35,13 @@ enum cip_flags { CIP_NONBLOCKING = 0x00, CIP_BLOCKING = 0x01, - CIP_SYNC_TO_DEVICE = 0x02, - CIP_EMPTY_WITH_TAG0 = 0x04, - CIP_DBC_IS_END_EVENT = 0x08, - CIP_WRONG_DBS = 0x10, - CIP_SKIP_DBC_ZERO_CHECK = 0x20, - CIP_SKIP_INIT_DBC_CHECK = 0x40, - CIP_EMPTY_HAS_WRONG_DBC = 0x80, - CIP_JUMBO_PAYLOAD = 0x100, + CIP_EMPTY_WITH_TAG0 = 0x02, + CIP_DBC_IS_END_EVENT = 0x04, + CIP_WRONG_DBS = 0x08, + CIP_SKIP_DBC_ZERO_CHECK = 0x10, + CIP_SKIP_INIT_DBC_CHECK = 0x20, + CIP_EMPTY_HAS_WRONG_DBC = 0x40, + CIP_JUMBO_PAYLOAD = 0x80, }; /** @@ -137,7 +134,6 @@ struct amdtp_stream { /* To wait for first packet. */ bool callbacked; wait_queue_head_t callback_wait; - struct amdtp_stream *sync_slave; /* For backends to process data blocks. */ void *protocol; @@ -223,23 +219,6 @@ static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) return sfc & 1; } -static inline void amdtp_stream_set_sync(enum cip_flags sync_mode, - struct amdtp_stream *master, - struct amdtp_stream *slave) -{ - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master->flags |= CIP_SYNC_TO_DEVICE; - slave->flags |= CIP_SYNC_TO_DEVICE; - master->sync_slave = slave; - } else { - master->flags &= ~CIP_SYNC_TO_DEVICE; - slave->flags &= ~CIP_SYNC_TO_DEVICE; - master->sync_slave = NULL; - } - - slave->sync_slave = NULL; -} - /** * amdtp_stream_wait_callback - sleep till callbacked or timeout * @s: the AMDTP stream -- cgit v0.10.2 From d9a16fc926a950c9e481cb7e89c554593b8e29e2 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:54 +0900 Subject: ALSA: firewire-lib: code cleanup for incoming packet handling In previous commit, this module has no need to reuse parameters of incoming packets for outgoing packets anymore. This commit arranges some needless codes for incoming packet processing. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 3468419..f1ebb7b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -447,17 +447,18 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks, unsigned int cycle, - unsigned int syt) + unsigned int payload_quadlets, unsigned int cycle) { + __be32 *buffer; u32 cip_header[2]; - unsigned int fmt, fdf; + unsigned int fmt, fdf, syt; unsigned int data_block_quadlets, data_block_counter, dbc_interval; + unsigned int data_blocks; struct snd_pcm_substream *pcm; unsigned int pcm_frames; bool lost; + buffer = s->buffer.packets[s->packet_index].buffer; cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); @@ -472,7 +473,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -483,7 +484,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Detect unexpected protocol: %08x %08x\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -492,7 +493,7 @@ static int handle_in_packet(struct amdtp_stream *s, fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT; if (payload_quadlets < 3 || (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) { - *data_blocks = 0; + data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -506,12 +507,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - *data_blocks = (payload_quadlets - 2) / data_block_quadlets; + data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -522,10 +523,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) + if (data_blocks > 0 && s->tx_dbc_interval > 0) dbc_interval = s->tx_dbc_interval; else - dbc_interval = *data_blocks; + dbc_interval = data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -538,13 +539,14 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); + syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; + pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + *data_blocks) & 0xff; + (data_block_counter + data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; @@ -618,10 +620,9 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int p, syt, packets; + unsigned int i, packets; unsigned int payload_quadlets, max_payload_quadlets; - unsigned int data_blocks; - __be32 *buffer, *headers = header; + __be32 *headers = header; u32 cycle; if (s->packet_index < 0) @@ -638,13 +639,12 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, /* For buffer-over-run prevention. */ max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4; - for (p = 0; p < packets; p++) { + for (i = 0; i < packets; i++) { cycle = increment_cycle_count(cycle, 1); - buffer = s->buffer.packets[s->packet_index].buffer; /* The number of quadlets in this packet */ payload_quadlets = - (be32_to_cpu(headers[p]) >> ISO_DATA_LENGTH_SHIFT) / 4; + (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT) / 4; if (payload_quadlets > max_payload_quadlets) { dev_err(&s->unit->device, "Detect jumbo payload: %02x %02x\n", @@ -652,14 +652,12 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, break; } - syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; - if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, cycle, syt) < 0) + if (handle_in_packet(s, payload_quadlets, cycle) < 0) break; } /* Queueing error or detecting invalid payload. */ - if (p < packets) { + if (i < packets) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; -- cgit v0.10.2 From 390a1512e6ccda2ec32ea1395814f36cf4d30e48 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:55 +0900 Subject: ALSA: firewire-lib: code cleanup for outgoing packet handling In previous commit, this module has no need to reuse parameters of incoming packets for outgoing packets anymore. This commit arranges some needless codes for outgoing packet processing. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f1ebb7b..6db2a73 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -411,15 +411,18 @@ static inline int queue_in_packet(struct amdtp_stream *s) amdtp_stream_get_max_payload(s), false); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, - unsigned int cycle, unsigned int syt) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) { __be32 *buffer; + unsigned int syt; + unsigned int data_blocks; unsigned int payload_length; unsigned int pcm_frames; struct snd_pcm_substream *pcm; buffer = s->buffer.packets[s->packet_index].buffer; + syt = calculate_syt(s, cycle); + data_blocks = calculate_data_blocks(s, syt); pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | @@ -588,8 +591,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int i, syt, packets = header_length / 4; - unsigned int data_blocks; + unsigned int i, packets = header_length / 4; u32 cycle; if (s->packet_index < 0) @@ -602,10 +604,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, for (i = 0; i < packets; ++i) { cycle = increment_cycle_count(cycle, 1); - syt = calculate_syt(s, cycle); - data_blocks = calculate_data_blocks(s, syt); - - if (handle_out_packet(s, data_blocks, cycle, syt) < 0) { + if (handle_out_packet(s, cycle) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; -- cgit v0.10.2 From 62f00e40b0718ebd8bd54fc7a9e89e873524d495 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 9 May 2016 23:15:56 +0900 Subject: ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag In former commit, drivers in ALSA firewire stack always starts IT context before IR context. If IR context starts after packets are transmitted by peer unit, packet discontinuity may be detected because the context starts in the middle of packet streaming. This situation is rare because IT context usually starts immediately. However, it's better to solve this issue. This is suppressed with CIP_SKIP_INIT_DBC_CHECK flag. This commit enables the same feature as CIP_SKIP_INIT_DBC_CHECK. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 6db2a73..830a95c 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -723,8 +723,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) goto err_unlock; } - if (s->direction == AMDTP_IN_STREAM && - s->flags & CIP_SKIP_INIT_DBC_CHECK) + if (s->direction == AMDTP_IN_STREAM) s->data_block_counter = UINT_MAX; else s->data_block_counter = 0; diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index da028b0..349c405 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -24,8 +24,6 @@ * The value of data_block_quadlets is used instead of reported value. * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. - * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first - * packet is not continuous from an initial value. * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty * packet is wrong but the others are correct. * @CIP_JUMBO_PAYLOAD: Only for in-stream. The number of data blocks in an @@ -39,9 +37,8 @@ enum cip_flags { CIP_DBC_IS_END_EVENT = 0x04, CIP_WRONG_DBS = 0x08, CIP_SKIP_DBC_ZERO_CHECK = 0x10, - CIP_SKIP_INIT_DBC_CHECK = 0x20, - CIP_EMPTY_HAS_WRONG_DBC = 0x40, - CIP_JUMBO_PAYLOAD = 0x80, + CIP_EMPTY_HAS_WRONG_DBC = 0x20, + CIP_JUMBO_PAYLOAD = 0x40, }; /** diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 0141813..4d3034a 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -526,8 +526,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) goto end; } - bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; - /* * BeBoB v3 transfers packets with these qurks: * - In the beginning of streaming, the value of dbc is incremented diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index 0ac92ab..b3cffd0 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -421,7 +421,7 @@ int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, /* Use different mode between incoming/outgoing. */ if (dir == AMDTP_IN_STREAM) { - flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK; + flags = CIP_NONBLOCKING; process_data_blocks = process_tx_data_blocks; } else { flags = CIP_BLOCKING; diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 7cb5743..d9361f3 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -242,8 +242,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * blocks than IEC 61883-6 defines. */ if (stream == &oxfw->tx_stream) { - oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | - CIP_JUMBO_PAYLOAD; + oxfw->tx_stream.flags |= CIP_JUMBO_PAYLOAD; if (oxfw->wrong_dbs) oxfw->tx_stream.flags |= CIP_WRONG_DBS; } -- cgit v0.10.2 From 8d879be882ba5a8bd4c2bc39bd2c336392564e13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 May 2016 16:07:40 +0200 Subject: ALSA: pcm: Bail out when chmap is already present When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a chmap has been already assigned, it returns as an error due to the conflicting snd_ctl_add() result. However, this also clears the already assigned chmap_kctl field via pcm_chmap_ctl_private_free(), and becomes inconsistent in the later operation. This patch adds the check of the conflicting chmap kctl before actually trying to allocate / assign. The check failure is treated as a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is basically a driver bug and having the stack trace would help developers to figure out the bad code path. Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0aca397..bb12615 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; -- cgit v0.10.2 From 420c470d6b5c2924a3182edf5b002870ff770331 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:29 +0100 Subject: ASoC: es8328: Move clock setup to hw_params This ensures that the clock is setup after its frequency has been set; the existing code in set_dai_fmt may be called before the clock rate has been set resulting in an incorrect configuration. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index afa6c5d..3ca89ae 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -445,9 +445,10 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; + int clk_rate = clk_get_rate(es8328->clk); int i; int reg; + int val; u8 ratio; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -455,16 +456,24 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, else reg = ES8328_ADCCONTROL5; - clk_rate = clk_get_rate(es8328->clk); - - if ((clk_rate != ES8328_SYSCLK_RATE_1X) && - (clk_rate != ES8328_SYSCLK_RATE_2X)) { + switch (clk_rate) { + case ES8328_SYSCLK_RATE_1X: + val = 0; + break; + case ES8328_SYSCLK_RATE_2X: + val = ES8328_MASTERMODE_MCLKDIV2; + break; + default: dev_err(codec->dev, "%s: clock is running at %d Hz, not %d or %d Hz\n", __func__, clk_rate, ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); return -EINVAL; } + ret = snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2, val); + if (ret < 0) + return ret; /* find master mode MCLK to sampling frequency ratio */ ratio = mclk_ratios[0].rate; @@ -484,8 +493,6 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; u8 mode = ES8328_DACCONTROL1_DACWL_16; /* set master/slave audio interface */ @@ -515,14 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_write(codec, ES8328_ADCCONTROL4, mode); /* Master serial port mode, with BCLK generated automatically */ - clk_rate = clk_get_rate(es8328->clk); - if (clk_rate == ES8328_SYSCLK_RATE_1X) - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MSC); - else - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MCLKDIV2 | - ES8328_MASTERMODE_MSC); + snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC); return 0; } -- cgit v0.10.2 From 57e41f3fb32a359753a3b2679c2502b2750bf6af Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:30 +0100 Subject: ASoC: es8328: Fix ADC format setup The ADCCONTROL4 and DACCONTROL1 registers are similar but not identical, with the DACCONTROL1 having each field starting one bit higher than ADCCONTROL4. Instead of introducing a magic shift, add new constants for the values in ADCCONTROL4 and use a second variable to setup the ADC. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3ca89ae..63e8262 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -493,7 +493,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u8 mode = ES8328_DACCONTROL1_DACWL_16; + u8 dac_mode = ES8328_DACCONTROL1_DACWL_16; + u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16; /* set master/slave audio interface */ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) @@ -502,13 +503,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S; break; case SND_SOC_DAIFMT_RIGHT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST; break; case SND_SOC_DAIFMT_LEFT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST; break; default: return -EINVAL; @@ -518,8 +522,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) return -EINVAL; - snd_soc_write(codec, ES8328_DACCONTROL1, mode); - snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode); + snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode); /* Master serial port mode, with BCLK generated automatically */ snd_soc_update_bits(codec, ES8328_MASTERMODE, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 156c748..5a4af01 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -84,7 +84,22 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL1 0x09 #define ES8328_ADCCONTROL2 0x0a #define ES8328_ADCCONTROL3 0x0b + #define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0) +#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2) +#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2) +#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2) +#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2) +#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5) + #define ES8328_ADCCONTROL5 0x0d #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) -- cgit v0.10.2 From 2da1ab667a506cc6a7dea88b70e6df3d281458f8 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:31 +0100 Subject: ASoC: es8328: Fix mask for VMIDSEL This is always used along with ES8328_CONTROL1_ENREF so there is no change in the generated code as a result of this fix. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 5a4af01..8bc79ff 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) -#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0) #define ES8328_CONTROL1_ENREF (1 << 2) #define ES8328_CONTROL1_SEQEN (1 << 3) #define ES8328_CONTROL1_SAMEFS (1 << 4) -- cgit v0.10.2 From f2ed04a4317e5c8074d98a5c1da175596811a2d8 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:32 +0100 Subject: ASoC: es8328: Use single R/W for regmap The chip only supports single reads and writes. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 63e8262..d580300 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -713,6 +713,7 @@ const struct regmap_config es8328_regmap_config = { .val_bits = 8, .max_register = ES8328_REG_MAX, .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, }; EXPORT_SYMBOL_GPL(es8328_regmap_config); -- cgit v0.10.2 From 8865c95e43257e6676bc0f6b042ecce17eff74fe Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:34 +0100 Subject: ASoC: es8328: Move sample size setup to hw_params This is a refactor in preparation for supporting more sample sizes which has no functional change. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index d580300..c5a36e6 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -482,9 +482,16 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, ratio = mclk_ratios[i].ratio; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACWL_MASK, + ES8328_DACCONTROL1_DACWL_16); + es8328->playback_fs = params_rate(params); es8328_set_deemph(codec); - } + } else + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCWL_MASK, + ES8328_ADCCONTROL4_ADCWL_16); return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); } @@ -493,8 +500,8 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u8 dac_mode = ES8328_DACCONTROL1_DACWL_16; - u8 adc_mode = ES8328_ADCCONTROL4_ADCWL_16; + u8 dac_mode = 0; + u8 adc_mode = 0; /* set master/slave audio interface */ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) @@ -522,8 +529,10 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) return -EINVAL; - snd_soc_write(codec, ES8328_DACCONTROL1, dac_mode); - snd_soc_write(codec, ES8328_ADCCONTROL4, adc_mode); + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode); + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode); /* Master serial port mode, with BCLK generated automatically */ snd_soc_update_bits(codec, ES8328_MASTERMODE, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 8bc79ff..9c33d8b 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -86,6 +86,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL3 0x0b #define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0) #define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0) #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0) #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0) @@ -95,6 +96,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2) #define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2) #define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2) +#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2) #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5) #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5) #define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5) @@ -124,6 +126,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL14 0x16 #define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1) #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) @@ -133,6 +136,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL1_DACWL_18 (2 << 3) #define ES8328_DACCONTROL1_DACWL_16 (3 << 3) #define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) -- cgit v0.10.2 From 779e86a31402c3f33f20bb02e99a5b75595bdf7f Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:35 +0100 Subject: ASoC: es8328: Support more sample formats The values are the same for the DAC and ADC so remove the specific values and use values with shifts. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index c5a36e6..a66c21c 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -60,7 +60,11 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = { #define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_11025) -#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) struct es8328_priv { struct regmap *regmap; @@ -449,6 +453,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, int i; int reg; int val; + int wl; u8 ratio; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -470,10 +475,28 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); return -EINVAL; } - ret = snd_soc_update_bits(codec, ES8328_MASTERMODE, + snd_soc_update_bits(codec, ES8328_MASTERMODE, ES8328_MASTERMODE_MCLKDIV2, val); - if (ret < 0) - return ret; + + switch (params_width(params)) { + case 16: + wl = 3; + break; + case 18: + wl = 2; + break; + case 20: + wl = 1; + break; + case 24: + wl = 0; + break; + case 32: + wl = 4; + break; + default: + return -EINVAL; + } /* find master mode MCLK to sampling frequency ratio */ ratio = mclk_ratios[0].rate; @@ -484,14 +507,14 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { snd_soc_update_bits(codec, ES8328_DACCONTROL1, ES8328_DACCONTROL1_DACWL_MASK, - ES8328_DACCONTROL1_DACWL_16); + wl << ES8328_DACCONTROL1_DACWL_SHIFT); es8328->playback_fs = params_rate(params); es8328_set_deemph(codec); } else snd_soc_update_bits(codec, ES8328_ADCCONTROL4, ES8328_ADCCONTROL4_ADCWL_MASK, - ES8328_ADCCONTROL4_ADCWL_16); + wl << ES8328_ADCCONTROL4_ADCWL_SHIFT); return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); } diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 9c33d8b..1a736e7 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -91,11 +91,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0) #define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0) #define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0) -#define ES8328_ADCCONTROL4_ADCWL_24 (0 << 2) -#define ES8328_ADCCONTROL4_ADCWL_20 (1 << 2) -#define ES8328_ADCCONTROL4_ADCWL_18 (2 << 2) -#define ES8328_ADCCONTROL4_ADCWL_16 (3 << 2) -#define ES8328_ADCCONTROL4_ADCWL_32 (4 << 2) +#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2 #define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2) #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5) #define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5) @@ -131,11 +127,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) -#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) -#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) -#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) -#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) -#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACWL_SHIFT 3 #define ES8328_DACCONTROL1_DACWL_MASK (7 << 3) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) -- cgit v0.10.2 From 45749c918129e409c44777f051dc0a5afb689459 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:36 +0100 Subject: ASoC: es8328: Support more sample rates Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index a66c21c..b8ca214 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -26,18 +26,30 @@ #include #include "es8328.h" -#define ES8328_SYSCLK_RATE_1X 11289600 -#define ES8328_SYSCLK_RATE_2X 22579200 +static const unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; -/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ -static struct { - int rate; - u8 ratio; -} mclk_ratios[] = { - { 8000, 9 }, - {11025, 7 }, - {22050, 4 }, - {44100, 2 }, +static const int ratios_12288[] = { + 10, 7, 6, 4, 3, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static const unsigned int rates_11289[] = { + 8018, 11025, 22050, 44100, 88200, +}; + +static const int ratios_11289[] = { + 9, 7, 4, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_11289 = { + .count = ARRAY_SIZE(rates_11289), + .list = rates_11289, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -57,9 +69,14 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = { "HPVDD", }; -#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ +#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_11025) + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_8000) #define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ @@ -71,6 +88,9 @@ struct es8328_priv { struct clk *clk; int playback_fs; bool deemph; + int mclkdiv2; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; + const int *mclk_ratios; struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; }; @@ -443,40 +463,55 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute) mute ? ES8328_DACCONTROL3_DACMUTE : 0); } +static int es8328_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->sysclk_constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + es8328->sysclk_constraints); + + return 0; +} + static int es8328_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate = clk_get_rate(es8328->clk); int i; int reg; - int val; int wl; - u8 ratio; + int ratio; + + if (!es8328->sysclk_constraints) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = ES8328_DACCONTROL2; else reg = ES8328_ADCCONTROL5; - switch (clk_rate) { - case ES8328_SYSCLK_RATE_1X: - val = 0; - break; - case ES8328_SYSCLK_RATE_2X: - val = ES8328_MASTERMODE_MCLKDIV2; - break; - default: - dev_err(codec->dev, - "%s: clock is running at %d Hz, not %d or %d Hz\n", - __func__, clk_rate, - ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + for (i = 0; i < es8328->sysclk_constraints->count; i++) + if (es8328->sysclk_constraints->list[i] == params_rate(params)) + break; + + if (i == es8328->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with current clock\n", + params_rate(params)); return -EINVAL; } + + ratio = es8328->mclk_ratios[i]; snd_soc_update_bits(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MCLKDIV2, val); + ES8328_MASTERMODE_MCLKDIV2, + es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0); switch (params_width(params)) { case 16: @@ -498,12 +533,6 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* find master mode MCLK to sampling frequency ratio */ - ratio = mclk_ratios[0].rate; - for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) - if (params_rate(params) <= mclk_ratios[i].rate) - ratio = mclk_ratios[i].ratio; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { snd_soc_update_bits(codec, ES8328_DACCONTROL1, ES8328_DACCONTROL1_DACWL_MASK, @@ -519,6 +548,40 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); } +static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int mclkdiv2 = 0; + + switch (freq) { + case 0: + es8328->sysclk_constraints = NULL; + es8328->mclk_ratios = NULL; + break; + case 22579200: + mclkdiv2 = 1; + /* fallthru */ + case 11289600: + es8328->sysclk_constraints = &constraints_11289; + es8328->mclk_ratios = ratios_11289; + break; + case 24576000: + mclkdiv2 = 1; + /* fallthru */ + case 12288000: + es8328->sysclk_constraints = &constraints_12288; + es8328->mclk_ratios = ratios_12288; + break; + default: + return -EINVAL; + } + + es8328->mclkdiv2 = mclkdiv2; + return 0; +} + static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -616,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec, } static const struct snd_soc_dai_ops es8328_dai_ops = { + .startup = es8328_startup, .hw_params = es8328_hw_params, .digital_mute = es8328_mute, + .set_sysclk = es8328_set_sysclk, .set_fmt = es8328_set_dai_fmt, }; -- cgit v0.10.2 From ca0d8797397c5daa6260a6c67b845d79f65140f5 Mon Sep 17 00:00:00 2001 From: John Keeping Date: Mon, 9 May 2016 12:24:37 +0100 Subject: ASoC: es8328: Set symmetric rates Although the ES8328 does support different rates for capture and playback, only very limited combinations are supported (8kHz and 48kHz or 8.0182kHz and 44.1kHz) with most rates required to be symmetric. Instead of adding a lot of complexity for little gain, let's enforce symmetric rates. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index b8ca214..2086d71 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -703,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = { .formats = ES8328_FORMATS, }, .ops = &es8328_dai_ops, + .symmetric_rates = 1, }; static int es8328_suspend(struct snd_soc_codec *codec) -- cgit v0.10.2 From fcc494af3cfaefc9f8a51c3c7e7f208a0553b28f Mon Sep 17 00:00:00 2001 From: Pardha Saradhi K Date: Tue, 10 May 2016 22:02:05 +0530 Subject: ASoC: Intel: Skylake: Add more SSP DAIs The Broxton-P platform has 6 SSPs so we need to add ssp2 thru ssp5 to DAI list for the driver. Signed-off-by: Pardha Saradhi K Signed-off-by: Ramesh Babu Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b0e7797..4494db6 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -770,6 +770,78 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { + .name = "SSP2 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp2 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp2 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP3 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp3 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp3 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP4 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp4 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp4 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP5 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp5 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp5 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ .name = "iDisp1 Pin", .ops = &skl_link_dai_ops, .playback = { -- cgit v0.10.2 From 76016322ec5670052fdabb08c586d6b16bd5062f Mon Sep 17 00:00:00 2001 From: Ramesh Babu Date: Tue, 10 May 2016 22:02:06 +0530 Subject: ASoC: Intel: Add Broxton-P machine driver This patch adds the Broxton-P machine driver for Intel Broxton-P reference boards. This machine uses the RT298 codec Signed-off-by: Ramesh Babu Signed-off-by: Senthilnathan Veppur Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 399afa1..91c15ab 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BXT_RT298_MACH + tristate "ASoC Audio driver for Broxton with RT298 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT298 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Broxton platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3310c0f..a850677 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c new file mode 100644 index 0000000..1b845ff --- /dev/null +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -0,0 +1,353 @@ +/* + * Intel Broxton-P I2S Machine Driver + * + * Copyright (C) 2014-2016, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Skylake I2S Machine driver + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt298.h" + +static struct snd_soc_jack broxton_headset; +/* Headset jack detection DAPM pins */ + +enum { + BXT_DPCM_AUDIO_PB = 0, + BXT_DPCM_AUDIO_CP, + BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_HDMI1_PB, + BXT_DPCM_AUDIO_HDMI2_PB, + BXT_DPCM_AUDIO_HDMI3_PB, +}; + +static struct snd_soc_jack_pin broxton_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broxton_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget broxton_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), +}; + +static const struct snd_soc_dapm_route broxton_rt298_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMic", NULL, "SoC DMIC"}, + + {"HDMI1", NULL, "hif5 Output"}, + {"HDMI2", NULL, "hif6 Output"}, + {"HDMI3", NULL, "hif7 Output"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp5 Tx"}, + { "ssp5 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp5 Rx" }, + { "ssp5 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, + +}; + +static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &broxton_headset, + broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins)); + + if (ret) + return ret; + + rt298_mic_detect(codec, &broxton_headset); + return 0; +} + +static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->codec_dai; + + return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id); +} + +static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP5 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int broxton_rt298_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broxton_rt298_ops = { + .hw_params = broxton_rt298_hw_params, +}; + +/* broxton digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broxton_rt298_dais[] = { + /* Front End DAI links */ + [BXT_DPCM_AUDIO_PB] + { + .name = "Bxt Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [BXT_DPCM_AUDIO_CP] + { + .name = "Bxt Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [BXT_DPCM_AUDIO_REF_CP] + { + .name = "Bxt Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI1_PB] + { + .name = "Bxt HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI2_PB] + { + .name = "Bxt HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI3_PB] + { + .name = "Bxt HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP5 - Codec */ + .name = "SSP5-Codec", + .be_id = 0, + .cpu_dai_name = "SSP5 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt298-aif1", + .init = broxton_rt298_codec_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broxton_ssp5_fixup, + .ops = &broxton_rt298_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .be_id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .be_id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .be_id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* broxton audio machine driver for SPT + RT298S */ +static struct snd_soc_card broxton_rt298 = { + .name = "broxton-rt298", + .owner = THIS_MODULE, + .dai_link = broxton_rt298_dais, + .num_links = ARRAY_SIZE(broxton_rt298_dais), + .controls = broxton_controls, + .num_controls = ARRAY_SIZE(broxton_controls), + .dapm_widgets = broxton_widgets, + .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), + .dapm_routes = broxton_rt298_map, + .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), + .fully_routed = true, +}; + +static int broxton_audio_probe(struct platform_device *pdev) +{ + broxton_rt298.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298); +} + +static struct platform_driver broxton_audio = { + .probe = broxton_audio_probe, + .driver = { + .name = "bxt_alc298s_i2s", + }, +}; +module_platform_driver(broxton_audio) + +/* Module information */ +MODULE_AUTHOR("Ramesh Babu "); +MODULE_AUTHOR("Senthilnathan Veppur "); +MODULE_DESCRIPTION("Intel SST Audio for Broxton"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bxt_alc298s_i2s"); -- cgit v0.10.2 From a0d5caeaebfd00853efa0080afc850e10be7b39a Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 10 May 2016 16:11:04 +0100 Subject: ASoC: da7213: Add DAI DAPM event to control DAI clocks Currently, when Codec is I2S master DAI clocks are continuously generated even if all audio streams have stopped. To improve efficiency, control of the DAI clocks for master mode have been moved to a DAPM widget event so they're only enabled as required. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 7278f93..701bd62 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -726,6 +726,36 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = { /* + * DAPM Events + */ + +static int da7213_dai_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable DAI clks for master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, + DA7213_DAI_CLK_EN_MASK); + return 0; + case SND_SOC_DAPM_POST_PMD: + /* Disable DAI clks if in master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* * DAPM widgets */ @@ -736,7 +766,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = { /* Use a supply here as this controls both input & output DAIs */ SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT, - DA7213_NO_INVERT, NULL, 0), + DA7213_NO_INVERT, da7213_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* * Input @@ -1143,11 +1174,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Set master/slave mode */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE; da7213->master = true; break; case SND_SOC_DAIFMT_CBS_CFS: - dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE; da7213->master = false; break; default: diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 030fd69..5de5c29 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -178,8 +178,6 @@ #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) -#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7) -#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ -- cgit v0.10.2 From d575b0b0f01a805508c5cf48b540f004e9b5de07 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 10 May 2016 16:11:05 +0100 Subject: ASoC: da7213: Add checking of SRM lock status before enabling DAI When the codec is DAI clk slave, and the SRM feature of the PLL is being used, the enabling of the DAI should occur only after the PLL has locked to the incoming WCLK. This update adds checking to the the DAI widget event, so it waits for SRM to lock. There is also a timeout if that lock doesn't occur within a given time. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 701bd62..680d111 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -734,6 +734,9 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -742,6 +745,26 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, DA7213_DAI_CLK_EN_MASK, DA7213_DAI_CLK_EN_MASK); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL); + if (!(pll_ctrl & DA7213_PLL_SRM_EN)) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7213_PLL_STATUS); + if (pll_status & DA7219_PLL_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + return 0; case SND_SOC_DAPM_POST_PMD: /* Disable DAI clks if in master mode */ diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 5de5c29..af75340 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -142,6 +142,9 @@ * Bit fields */ +/* DA7213_PLL_STATUS = 0x03 */ +#define DA7219_PLL_SRM_LOCK (0x1 << 1) + /* DA7213_SR = 0x22 */ #define DA7213_SR_8000 (0x1 << 0) #define DA7213_SR_11025 (0x2 << 0) @@ -502,6 +505,7 @@ #define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 #define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 #define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7213_SRM_CHECK_RETRIES 8 enum da7213_clk_src { DA7213_CLKSRC_MCLK = 0, -- cgit v0.10.2 From 7e28fd469624fc41ec326a31abbc63a7afdd10f5 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 10 May 2016 16:11:06 +0100 Subject: ASoC: da7213: Default PC counter to free-running when DAI disabled Currently PC counter is always synchronised to DAI which means that when the DAI is disabled, features such as ALC calibration cannot be executed successfully. This patch makes sure that when the DAI is disabled, PC counter is set to free-running. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 680d111..657b7eb 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -746,6 +746,10 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, DA7213_DAI_CLK_EN_MASK, DA7213_DAI_CLK_EN_MASK); + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, 0); + /* Slave mode, if SRM not enabled no need for status checks */ pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL); if (!(pll_ctrl & DA7213_PLL_SRM_EN)) @@ -767,6 +771,11 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, return 0; case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + /* Disable DAI clks if in master mode */ if (da7213->master) snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, @@ -1599,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec) /* Default to using SRM for slave mode */ da7213->srm_en = true; + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA7213_AUX_L_CTRL, DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index af75340..26b87e3c 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -413,6 +413,9 @@ #define DA7213_DMIC_CLK_RATE_SHIFT 2 #define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2) +/* DA7213_PC_COUNT = 0x94 */ +#define DA7213_PC_FREERUN_MASK (0x1 << 0) + /* DA7213_DIG_CTRL = 0x99 */ #define DA7213_DAC_L_INV_SHIFT 3 #define DA7213_DAC_R_INV_SHIFT 7 -- cgit v0.10.2 From 1e62c52ddc2d23a02ac2308cc1bb6ff18f0cf3cd Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 10 May 2016 16:11:07 +0100 Subject: ASoC: da7213: Update PLL ranges to improve locking at frequency boundary This update changes the dividers used for ranges of input MCLK frequencies, to improve PLL locking for a corner case when at edge of MCLK frequency input divider range. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 657b7eb..a233fe7 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1344,26 +1344,26 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Workout input divider based on MCLK rate */ if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { /* 32KHz PLL Mode */ - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; freq_ref = 3750000; pll_ctrl |= DA7213_PLL_32K_MODE; } else { /* 5 - 54MHz MCLK */ if (da7213->mclk_rate < 5000000) { goto pll_err; - } else if (da7213->mclk_rate <= 10000000) { - indiv_bits = DA7213_PLL_INDIV_5_10_MHZ; - indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7213->mclk_rate <= 20000000) { - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7213->mclk_rate <= 40000000) { - indiv_bits = DA7213_PLL_INDIV_20_40_MHZ; - indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7213->mclk_rate <= 9000000) { + indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ; + indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL; + } else if (da7213->mclk_rate <= 18000000) { + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7213->mclk_rate <= 36000000) { + indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7213->mclk_rate <= 54000000) { - indiv_bits = DA7213_PLL_INDIV_40_54_MHZ; - indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL; } else { goto pll_err; } diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 26b87e3c..fbb7a35 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -163,10 +163,10 @@ #define DA7213_VMID_EN (0x1 << 7) /* DA7213_PLL_CTRL = 0x27 */ -#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2) -#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2) -#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2) -#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2) +#define DA7213_PLL_INDIV_5_TO_9_MHZ (0x0 << 2) +#define DA7213_PLL_INDIV_9_TO_18_MHZ (0x1 << 2) +#define DA7213_PLL_INDIV_18_TO_36_MHZ (0x2 << 2) +#define DA7213_PLL_INDIV_36_TO_54_MHZ (0x3 << 2) #define DA7213_PLL_INDIV_MASK (0x3 << 2) #define DA7213_PLL_MCLK_SQR_EN (0x1 << 4) #define DA7213_PLL_32K_MODE (0x1 << 5) @@ -499,16 +499,16 @@ #define DA7213_ALC_AVG_ITERATIONS 5 /* PLL related */ -#define DA7213_SYSCLK_MCLK 0 -#define DA7213_SYSCLK_PLL 1 -#define DA7213_PLL_FREQ_OUT_90316800 90316800 -#define DA7213_PLL_FREQ_OUT_98304000 98304000 -#define DA7213_PLL_FREQ_OUT_94310400 94310400 -#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 -#define DA7213_SRM_CHECK_RETRIES 8 +#define DA7213_SYSCLK_MCLK 0 +#define DA7213_SYSCLK_PLL 1 +#define DA7213_PLL_FREQ_OUT_90316800 90316800 +#define DA7213_PLL_FREQ_OUT_98304000 98304000 +#define DA7213_PLL_FREQ_OUT_94310400 94310400 +#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL 2 +#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL 16 +#define DA7213_SRM_CHECK_RETRIES 8 enum da7213_clk_src { DA7213_CLKSRC_MCLK = 0, -- cgit v0.10.2 From abc189eadf6c12e60f95030e9c84083175526eaf Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 10 May 2016 16:11:08 +0100 Subject: ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk Current checking for PLL 32KHz mode fails in driver code when bypassing the PLL. This is due to an incorrect check of PLL source type when 32KHz clock is provided. Removal of this check resolves the issue. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index a233fe7..e5527bc 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1342,7 +1342,7 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_ctrl = 0; /* Workout input divider based on MCLK rate */ - if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { + if (da7213->mclk_rate == 32768) { /* 32KHz PLL Mode */ indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; -- cgit v0.10.2 From 396cbebeeb9734aee8efe39431d3b96655bf1e94 Mon Sep 17 00:00:00 2001 From: Joonas Lahtinen Date: Tue, 10 May 2016 09:08:57 +0300 Subject: ASoC: Intel: Fix printk formatting Format number after 0x in hex. Cc: Jie Yang Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Signed-off-by: Joonas Lahtinen Signed-off-by: Mark Brown diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c..994256b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { -- cgit v0.10.2 From bfb7802a06ac1855096a3f248822e8f943e6574d Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Wed, 11 May 2016 11:07:02 +1000 Subject: ASoC: Intel: fix up for DAI link's be_id change Signed-off-by: Stephen Rothwell Acked-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 1b845ff..f478751 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -254,7 +254,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { { /* SSP5 - Codec */ .name = "SSP5-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP5 Pin", .platform_name = "0000:00:0e.0", .no_pcm = 1, @@ -271,7 +271,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { }, { .name = "dmic01", - .be_id = 1, + .id = 1, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -282,7 +282,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -293,7 +293,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -304,7 +304,7 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", -- cgit v0.10.2 From 1dba9db0eaa64d362d9d9afb5eeaececdaef948d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 12 May 2016 02:17:39 +0900 Subject: ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity These three commits were merged to improve PCM pointer granularity. commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call") commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position") commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing") The point of them is to handle queued packets not only in software IRQ context of IR/IT contexts, but also in process context. As a result of handling packets, period tasklet is scheduled when acrossing PCM period boundary. This is to prevent recursive call of 'struct snd_pcm_ops.pointer()' in the same context. When the pointer callback is executed in the process context, it's better to avoid the second callback in the software IRQ context. The software IRQ context runs immediately after scheduled in the process context because few packets are queued yet. For the aim, 'pointer_flush' is used, however it causes a race condition between the process context and software IRQ context of IR/IT contexts. Practically, this race is not so critical because it influences process context to skip flushing queued packet and to get worse granularity of PCM pointer. The race condition is quite rare but it should be improved for stable service. The similar effect can be achieved by using 'in_interrupt()' macro. This commit obsoletes 'pointer_flush' with it. Acked-by: Clemens Ladisch Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 830a95c..024ab7f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -251,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -356,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -803,11 +801,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets will be already + * queued in the buffer. These packets should be handled immediately + * to keep better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 349c405..c1bc7fa 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -126,7 +126,6 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; -- cgit v0.10.2 From a9c4284bf5a95c4788e7fbf3c46b14dcbfda3a6d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 11 May 2016 07:33:27 +0900 Subject: ALSA: firewire-lib: add context information to tracepoints In current implementation, packet processing is done in both of software IRQ contexts of IR/IT contexts and process contexts. This is usual interrupt handling of IR/IT context for 1394 OHCI. (in hardware IRQ context) irq_handler() (drivers/firewire/ohci.c) ->tasklet_schedule() (in software IRQ context) handle_it_packet() or handle_ir_packet_per_buffer() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() However, we have another chance for packet processing. It's done in PCM frame handling via ALSA PCM interfaces. (in process context) ioctl(i.e. SNDRV_PCM_IOCTL_HWSYNC) ->snd_pcm_hwsync() (sound/core/pcm_native.c) ->snd_pcm_update_hw_ptr() (sound/core/pcm_lib.c) ->snd_pcm_update_hw_ptr0() ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() (drivers/firewire/core-iso.c) ->struct fw_card_driver.flush_iso_completions() = ohci_flush_iso_completions() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() This design is for a better granularity of PCM pointer. When ioctl(2) is executed with some commands for ALSA PCM interface, queued packets are handled at first. Then, the latest number of handled PCM frames is reported. The number can represent PCM frames transferred in most near isochronous cycle. Current tracepoints include no information to distinguish running contexts. When tracing the interval of software IRQ context, this is not good. This commit adds more information for current context. Additionally, the index of packet processed in one context is added in a case that packet processing is executed in continuous context of the same kind, As a result, the output includes 11 fields with additional two fields to commit 0c95c1d6197f ("ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data"): 17131.9186: out_packet: 07 7494 ffc0 ffc1 00 000700c0 9001a496 058 45 1 13 17131.9186: out_packet: 07 7495 ffc0 ffc1 00 000700c8 9001ba00 058 46 1 14 17131.9186: out_packet: 07 7496 ffc0 ffc1 00 000700d0 9001ffff 002 47 1 15 17131.9189: out_packet: 07 7497 ffc0 ffc1 00 000700d0 9001d36a 058 00 0 00 17131.9189: out_packet: 07 7498 ffc0 ffc1 00 000700d8 9001e8d4 058 01 0 01 17131.9189: out_packet: 07 7499 ffc0 ffc1 00 000700e0 9001023e 058 02 0 00 17131.9206: in_packet: 07 7447 ffc1 ffc0 01 3f070072 9001783d 058 32 1 00 17131.9206: in_packet: 07 7448 ffc1 ffc0 01 3f070072 90ffffff 002 33 1 01 17131.9206: in_packet: 07 7449 ffc1 ffc0 01 3f07007a 900191a8 058 34 1 02 (Here, some common fields are omitted so that a line is within 80 characters.) The legend is: - The second of cycle scheduled for the packet - The count of cycle scheduled for the packet - The ID of node as source (hex) - The ID of node as destination (hex) - The value of isochronous channel - The first quadlet of CIP header (hex) - The second quadlet of CIP header (hex) - The number of included quadlets - The index of packet in a buffer maintained by this module - 0 in process context, 1 in IRQ context - The index of packet processed in the context Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 425d1d7..1622579 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -14,8 +14,8 @@ #include TRACE_EVENT(in_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets), - TP_ARGS(s, cycles, cip_header, payload_quadlets), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -25,6 +25,8 @@ TRACE_EVENT(in_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -36,10 +38,12 @@ TRACE_EVENT(in_packet, __entry->cip_header0 = cip_header[0]; __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -48,12 +52,14 @@ TRACE_EVENT(in_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); TRACE_EVENT(out_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length), - TP_ARGS(s, cycles, cip_header, payload_length), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -63,6 +69,8 @@ TRACE_EVENT(out_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -74,10 +82,12 @@ TRACE_EVENT(out_packet, __entry->cip_header0 = be32_to_cpu(cip_header[0]); __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -86,6 +96,8 @@ TRACE_EVENT(out_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 024ab7f..bf10ca3 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -409,7 +409,8 @@ static inline int queue_in_packet(struct amdtp_stream *s) amdtp_stream_get_max_payload(s), false); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; unsigned int syt; @@ -434,7 +435,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - trace_out_packet(s, cycle, buffer, payload_length); + trace_out_packet(s, cycle, buffer, payload_length, index); if (queue_out_packet(s, payload_length, false) < 0) return -EIO; @@ -448,7 +449,8 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, unsigned int cycle) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { __be32 *buffer; u32 cip_header[2]; @@ -463,7 +465,7 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); - trace_in_packet(s, cycle, cip_header, payload_quadlets); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); /* * This module supports 'Two-quadlet CIP header with SYT field'. @@ -602,7 +604,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, for (i = 0; i < packets; ++i) { cycle = increment_cycle_count(cycle, 1); - if (handle_out_packet(s, cycle) < 0) { + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -649,7 +651,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, break; } - if (handle_in_packet(s, payload_quadlets, cycle) < 0) + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; } -- cgit v0.10.2 From ff38e0c70adce96de0be0bf470cbb9e283ba6965 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 11 May 2016 07:35:09 +0900 Subject: ALSA: firewire-lib: drop skip argument from helper functions to queue a packet On most of audio and music units on IEEE 1394 bus which ALSA firewire stack supports (or plans to support), CIP with two quadlets header is used. Thus, there's no cases to queue packets with blank payload. If such packets are going to be queued, it means that they're for skips of the cycle. This commit simplifies helper functions to queue a packet. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index bf10ca3..00060c4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -368,9 +368,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -381,8 +380,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -397,16 +398,15 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, @@ -437,7 +437,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, trace_out_packet(s, cycle, buffer, payload_length, index); - if (queue_out_packet(s, payload_length, false) < 0) + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -764,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); -- cgit v0.10.2 From 32902177f7f6ae70e1d5e71d935aa1bfcae7f01c Mon Sep 17 00:00:00 2001 From: John Keeping Date: Thu, 12 May 2016 13:55:53 +0100 Subject: ASoC: dapm: deprecate MICBIAS widget type Commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget", 2011-09-23) says: A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs. but this has never been documented as a policy. Add some comments to make it clear. Signed-off-by: John Keeping Signed-off-by: Mark Brown diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..3101d53 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -100,6 +100,7 @@ struct device; { .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} +/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */ #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_micbias, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ @@ -473,7 +474,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_out_drv, /* output driver */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ - snd_soc_dapm_micbias, /* microphone bias (power) */ + snd_soc_dapm_micbias, /* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */ snd_soc_dapm_mic, /* microphone */ snd_soc_dapm_hp, /* headphones */ snd_soc_dapm_spk, /* speaker */ -- cgit v0.10.2 From bb7cb54b388d8d0fbb3af27f14b121ee9c92e867 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 09:38:49 +0530 Subject: ASoC: rt298: fix null deref on acpi driver data ACPI driver data can be NULL so we need to check that before dereference the driver data. Signed-off-by: Senthilnathan Veppur Signed-off-by: Vinod Koul Acked-by: Bard Liao Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f0e6c06..52aacb1 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1184,7 +1184,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } -- cgit v0.10.2 From b9c17f13ba484d8492278c67cd95b7207def776f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 09:38:50 +0530 Subject: ASoC: rt298: Add DMI match for Broxton-P reference platform Broxton-P reference platform also uses combo jack for audio connector so we need to set codec pdata to use this based on DMI match for this board. Signed-off-by: Ramesh Babu Signed-off-by: Senthilnathan Veppur Signed-off-by: Vinod Koul Acked-by: Bard Liao Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 52aacb1..a1aaffc 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -1132,6 +1133,17 @@ static const struct acpi_device_id rt298_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); +static const struct dmi_system_id force_combo_jack_table[] = { + { + .ident = "Intel Broxton P", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"), + DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P") + } + }, + { } +}; + static int rt298_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1189,6 +1201,11 @@ static int rt298_i2c_probe(struct i2c_client *i2c, acpiid->driver_data; } + if (dmi_check_system(force_combo_jack_table)) { + rt298->pdata.cbj_en = true; + rt298->pdata.gpio2_en = false; + } + /* VREF Charging */ regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80); regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860); -- cgit v0.10.2 From 639db596165746ca87bbcb56559b094fd9042890 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 May 2016 18:04:16 +0200 Subject: ALSA: au88x0: Fix zero clear of stream->resources There are a few calls of memset() to stream->resources, but they all are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST, while this field is a u32 array. This may leave the memories not zero-cleared. Fix it by replacing them with a simpler sizeof(stream->resources) instead. Reported-by: David Binderman Signed-off-by: Takashi Iwai diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c32..4a054d7 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } -- cgit v0.10.2 From 4446085d21e75dd6c0c45577f12db0bd7c7bf35f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 08:58:53 +0530 Subject: ALSA: hdac: add link pm and ref counting The HDA links can be switched off when not is use, similarly command DMA can be stopped as well. This calls for a reference counting mechanism on the link by it's users to manage the link power. The DMA can be turned off when all links are off For this we add two APIs snd_hdac_ext_bus_link_get snd_hdac_ext_bus_link_put They help users to turn up/down link and manage the DMA as well Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 07fa592..b9593b2 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -14,6 +14,8 @@ * @gtscap: gts capabilities pointer * @drsmcap: dma resume capabilities pointer * @hlink_list: link list of HDA links + * @lock: lock for link mgmt + * @cmd_dma_state: state of cmd DMAs: CORB and RIRB */ struct hdac_ext_bus { struct hdac_bus bus; @@ -27,6 +29,9 @@ struct hdac_ext_bus { void __iomem *drsmcap; struct list_head hlink_list; + + struct mutex lock; + bool cmd_dma_state; }; int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev, @@ -142,6 +147,9 @@ struct hdac_ext_link { void __iomem *ml_addr; /* link output stream reg pointer */ u32 lcaps; /* link capablities */ u16 lsdiid; /* link sdi identifier */ + + int ref_count; + struct list_head list; }; @@ -154,6 +162,11 @@ void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream); +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link); + /* update register macro */ #define snd_hdac_updatel(addr, reg, mask, val) \ writel(((readl(addr + reg) & ~(mask)) | (val)), \ diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c..3b7ae24 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, INIT_LIST_HEAD(&ebus->hlink_list); ebus->idx = idx++; + mutex_init(&ebus->lock); + ebus->cmd_dma_state = true; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 548cc1e..860f8ca 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP); hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID); + /* since link in On, update the ref */ + hlink->ref_count = 1; + list_add_tail(&hlink->list, &ebus->hlink_list); } @@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); + +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + + mutex_lock(&ebus->lock); + + /* + * if we move from 0 to 1, count will be 1 so power up this link + * as well, also check the dma status and trigger that + */ + if (++link->ref_count == 1) { + if (!ebus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(&ebus->bus); + ebus->cmd_dma_state = true; + } + + ret = snd_hdac_ext_bus_link_power_up(link); + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); + +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + struct hdac_ext_link *hlink; + bool link_up = false; + + mutex_lock(&ebus->lock); + + /* + * if we move from 1 to 0, count will be 0 + * so power down this link as well + */ + if (--link->ref_count == 0) { + ret = snd_hdac_ext_bus_link_power_down(link); + + /* + * now check if all links are off, if so turn off + * cmd dma as well + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) { + link_up = true; + break; + } + } + + if (!link_up) { + snd_hdac_bus_stop_cmd_io(&ebus->bus); + ebus->cmd_dma_state = false; + } + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); -- cgit v0.10.2 From cce6c149eba3aabf678ffea91ac1e4103b9c185e Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 08:58:54 +0530 Subject: ASoC: Intel: Skylake: add link management Use shiny new link APIs to manage the links. Also remove old link configuration logic from driver. We need to keep link and cmd dma to off during active suspend to allow system to enter low power state and turn it on if the link and cmd dma was on before active suspend in active resume. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 4494db6..1548023 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -533,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, if (!link) return -EINVAL; - snd_hdac_ext_bus_link_power_up(link); snd_hdac_ext_link_stream_reset(link_dev); snd_hdac_ext_link_stream_setup(link_dev, format_val); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 83e985c..06d8c26 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -229,7 +229,12 @@ static int skl_suspend(struct device *dev) * running, we need to save the state for these and continue */ if (skl->supend_active) { + /* turn off the links and stop the CORB/RIRB DMA if it is On */ snd_hdac_ext_bus_link_power_down_all(ebus); + + if (ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); + enable_irq_wake(bus->irq); pci_save_state(pci); pci_disable_device(pci); @@ -255,6 +260,7 @@ static int skl_resume(struct device *dev) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; int ret; /* Turned OFF in HDMI codec driver after codec reconfiguration */ @@ -276,8 +282,29 @@ static int skl_resume(struct device *dev) ret = pci_enable_device(pci); snd_hdac_ext_bus_link_power_up_all(ebus); disable_irq_wake(bus->irq); + /* + * turn On the links which are On before active suspend + * and start the CORB/RIRB DMA if On before + * active suspend. + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) + snd_hdac_ext_bus_link_power_up(hlink); + } + + if (ebus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(&ebus->bus); } else { ret = _skl_resume(ebus); + + /* turn off the links which are off before suspend */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (!hlink->ref_count) + snd_hdac_ext_bus_link_power_down(hlink); + } + + if (!ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); } return ret; @@ -613,6 +640,7 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; + struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -679,6 +707,12 @@ static int skl_probe(struct pci_dev *pci, } } + /* + * we are done probling so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + /*configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); -- cgit v0.10.2 From b2047e996cd88d36eb0f4e84fe6aedab831a4b31 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 08:58:55 +0530 Subject: ASoC: hdac_hdmi: add link management Manage the hda idisp link using shiny new link APIs. We need to keep link On while we probe and also hold the reference in runtime resume and drop in suspend Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index aaa038f..13002f3 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1378,10 +1378,18 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_pin *pin; + struct hdac_ext_link *hlink = NULL; int ret; edev->scodec = codec; + /* + * hold the ref while we probe, also no need to drop the ref on + * exit, we call pm_runtime_suspend() so that will do for us + */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; @@ -1480,9 +1488,14 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) struct hdac_device *codec = &edev->hdac; struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; + struct hdac_ext_link *hlink = NULL; int num_dais = 0; int ret = 0; + /* hold the ref while we probe */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; @@ -1516,8 +1529,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) } /* ASoC specific initialization */ - return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, - hdmi_dais, num_dais); + ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, + hdmi_dais, num_dais); + + snd_hdac_ext_bus_link_put(edev->ebus, hlink); + + return ret; } static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) @@ -1556,6 +1573,9 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + unsigned long timeout; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1579,6 +1599,9 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return err; } + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_put(ebus, hlink); + return 0; } @@ -1587,6 +1610,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1595,6 +1620,9 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_get(ebus, hlink); + err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); -- cgit v0.10.2 From 97d3ddd71fbf663a5da52897757333341a8b254f Mon Sep 17 00:00:00 2001 From: Florian Meier Date: Fri, 13 May 2016 09:14:12 +0000 Subject: ASoC: pcm5102a: Add support for PCM5102A codec Some definitions to support the PCM5102A codec by Texas Instruments. Signed-off-by: Florian Meier Changes to original patch by Florian Meier: * rebased (Makefile and Kconfig * fixed checkpath errors (spaces, newlines) * added dt-binding documentation Signed-off-by: Martin Sperl Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 0000000..c63ab0b6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODECs + +These devices does not use I2C or SPI. + +Required properties: + + - compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..f736953 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -94,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER + select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C @@ -575,6 +576,9 @@ config SND_SOC_PCM3168A_SPI select SND_SOC_PCM3168A select REGMAP_SPI +config SND_SOC_PCM5102A + tristate + config SND_SOC_PCM512x tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712..4532a74 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -89,6 +89,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o +snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o @@ -298,6 +299,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o +obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c new file mode 100644 index 0000000..ed51567 --- /dev/null +++ b/sound/soc/codecs/pcm5102a.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM5102A codec + * + * Author: Florian Meier + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include + +#include + +static struct snd_soc_dai_driver pcm5102a_dai = { + .name = "pcm5102a-hifi", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; + +static int pcm5102a_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a, + &pcm5102a_dai, 1); +} + +static int pcm5102a_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id pcm5102a_of_match[] = { + { .compatible = "ti,pcm5102a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm5102a_of_match); + +static struct platform_driver pcm5102a_codec_driver = { + .probe = pcm5102a_probe, + .remove = pcm5102a_remove, + .driver = { + .name = "pcm5102a-codec", + .owner = THIS_MODULE, + .of_match_table = pcm5102a_of_match, + }, +}; + +module_platform_driver(pcm5102a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM5102A codec driver"); +MODULE_AUTHOR("Florian Meier "); +MODULE_LICENSE("GPL v2"); -- cgit v0.10.2 From 48a260eec301fd7a112d1737ca2755d91558a349 Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:13:55 +0200 Subject: ASoC: fsl_ssi: Real hardware channels max number is 32 The max number of slots in TDM mode is 32: - Frame Rate Divider Control is a 5bit value - Time slot mask registers control 32 slots. Signed-off-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de10..8d5f3c19 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1158,14 +1158,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, -- cgit v0.10.2 From e09745f2e6a1f692fc63db01850aacf025475aad Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:13:56 +0200 Subject: ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. im6sl reference manual 47.7.4: " Bit clock - Used to serially clock the data bits in and out of the SSI port. This clock is either generated internally (from SSI's sys clock) or taken from external clock source (through the Tx/Rx clock ports). [...] Care should be taken to ensure that the bit clock frequency (either internally generated by dividing the SSI's sys clock or sourced from external device through Tx/Rx clock ports) is never greater than 1/5 of the ipg_clk (from CCM) frequency. " Since, in master mode, the sysclk is a multiple of bitclk, we can easily reach a high sysclk value, whereas keeping a reasonable bitclk. ex: 8ch x 16bit x 48kHz = 6144000, requires a 24576000 sysclk (PM=1) yet ipg_clk/5 = 66Mhz/5 = 13.2 Signed-off-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8d5f3c19..86229c8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -670,6 +670,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (IS_ERR(ssi_private->baudclk)) return -EINVAL; + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (freq * 5 > clk_get_rate(ssi_private->clk)) { + dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n"); + return -EINVAL; + } + baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream)); /* It should be already enough to divide clock by setting pm alone */ @@ -686,13 +695,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - /* - * Hardware limitation: The bclk rate must be - * never greater than 1/5 IPG clock rate - */ - if (clkrate * 5 > clk_get_rate(ssi_private->clk)) - continue; - clkrate /= factor; afreq = clkrate / (i + 1); -- cgit v0.10.2 From 0096b693962d3abde4f41b13cf03c765f33e9d8d Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:13:57 +0200 Subject: ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. Most of functions only receive the ssi_private reference and don't have a knowledge of 'dev' pointer, even for debug purpose. Signed-off-by: Arnaud Mouiche Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 86229c8..149df3c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -261,6 +261,7 @@ struct fsl_ssi_private { struct fsl_ssi_dbg dbg_stats; const struct fsl_ssi_soc_data *soc; + struct device *dev; }; /* @@ -1404,6 +1405,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->soc = of_id->data; + ssi_private->dev = &pdev->dev; sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { -- cgit v0.10.2 From d9f2a202877c15818d98268f47d6b4bcfcb84437 Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:13:58 +0200 Subject: ASoC: fsl_ssi: Fix samples being dropped at Playback startup If the capture is already running while playback is started, it is highly probable (>80% in a 8 channels scenario) that samples are lost between the DMA and TX fifo. The reason is that SIER.TDMAE is set before STCR.TFEN0, leaving a time window where the FIFO doesn't receive the samples written by the DMA. This particular case happened only if capture is already enabled as SCR.SSIEN is already set at the playback startup instant. Signed-off-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 149df3c..47ebb83 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -475,9 +475,9 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, * (online configuration) */ if (enable) { - regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr); + regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); } else { u32 sier; u32 srcr; -- cgit v0.10.2 From 61fcf10a0ee44763e0347b297a377137f8950772 Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:13:59 +0200 Subject: ASoC: fsl_ssi: Fix channel slipping in Playback at startup Previously, SCR.SSIEN and SCR.TE were enabled at once if no capture stream was also running. This may not give a chance for the DMA to write the first sample in TX FIFO before the streaming starts on the PCM bus, inserting void samples first. Those void samples are then responsible for slipping the channels. Signed-off-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 47ebb83..8944af5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -507,8 +507,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, config_done: /* Enabling of subunits is done after configuration */ - if (enable) + if (enable) { + if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) { + /* + * Be sure the Tx FIFO is filled when TE is set. + * Otherwise, there are some chances to start the + * playback with some void samples inserted first, + * generating a channel slip. + * + * First, SSIEN must be set, to let the FIFO be filled. + * + * Notes: + * - Limit this fix to the DMA case until FIQ cases can + * be tested. + * - Limit the length of the busy loop to not lock the + * system too long, even if 1-2 loops are sufficient + * in general. + */ + int i; + int max_loop = 100; + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN); + for (i = 0; i < max_loop; i++) { + u32 sfcsr; + regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr); + if (CCSR_SSI_SFCSR_TFCNT0(sfcsr)) + break; + } + if (i == max_loop) { + dev_err(ssi_private->dev, + "Timeout waiting TX FIFO filling\n"); + } + } regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr); + } } -- cgit v0.10.2 From 027db2e122db81b055a2b569d72f2f1d29c4d007 Mon Sep 17 00:00:00 2001 From: Arnaud Mouiche Date: Tue, 3 May 2016 14:14:00 +0200 Subject: ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. Happened when the Playback (or Capture) is running continuously and Capture (or Playback) is restarted (xrun, manual stop/start...) Since the RX (or TX) FIFO are only reset when the whole SSI is disabled, pending samples from previous capture (or playback) session may still be present. They must be erased to not introduce channel slipping. FIFO Clear register fields are documented in IMX51, IMX35 reference manual. They are not documented in IMX50 or IMX6 RM, despite they are working as expected on IMX6SL and IMX6solo. Signed-off-by: Arnaud Mouiche Reviewed-by: Fabio Estevam Tested-by: Caleb Crome Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8944af5..d2dd47d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -401,6 +401,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private, } /* + * Clear RX or TX FIFO to remove samples from the previous + * stream session which may be still present in the FIFO and + * may introduce bad samples and/or channel slipping. + * + * Note: The SOR is not documented in recent IMX datasheet, but + * is described in IMX51 reference manual at section 56.3.3.15. + */ +static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private, + bool is_rx) +{ + if (is_rx) { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR); + } else { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR); + } +} + +/* * Calculate the bits that have to be disabled for the current stream that is * getting disabled. This keeps the bits enabled that are necessary for the * second stream to work if 'stream_active' is true. @@ -475,6 +495,8 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, * (online configuration) */ if (enable) { + fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE); + regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr); regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); -- cgit v0.10.2 From 29cf67b99527296e9c9b9f6995c1d271d21f79c0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 16 May 2016 09:50:07 -0300 Subject: ASoC: hdac_hdmi: Remove the unused 'timeout' variable Commit b2047e996cd88d3 ("ASoC: hdac_hdmi: add link management") introuduced the following build warning: sound/soc/codecs/hdac_hdmi.c:1721:16: warning: unused variable 'timeout' [-Wunused-variable] Remove the unused 'timeout' variable. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3733297..181cd3b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1718,7 +1718,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; - unsigned long timeout; struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; -- cgit v0.10.2 From c7c5856b6f6f30de622bf87947363890b286553f Mon Sep 17 00:00:00 2001 From: Muhammad Falak R Wani Date: Sun, 15 May 2016 16:26:51 +0530 Subject: sound: oss: Use setup_timer and mod_timer. The function setup_timer combines the initialization of a timer with the initialization of the timer's function and data fields. The mulitiline code for timer initialization is now replaced with function setup_timer. Also, quoting the mod_timer() function comment: -> mod_timer() is a more efficient way to update the expire field of an active timer (if the timer is inactive it will be activated). Use setup_timer() and mod_timer() to setup and arm a timer, making the code compact and aid readablity. Signed-off-by: Muhammad Falak R Wani Signed-off-by: Takashi Iwai diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index b36ea47..0b8d0de 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - init_timer(&vnc_timer); - vnc_timer.function = vnc_slider_tick; - vnc_timer.expires = jiffies; - vnc_timer.data = nr_waveartist_devs; - add_timer(&vnc_timer); + setup_timer(&vnc_timer, vnc_slider_tick, + nr_waveartist_devs); + mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); -- cgit v0.10.2 From 17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 18 May 2016 22:27:43 +0900 Subject: ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type Commit a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") adds new members to tracepoint events of this module, to represent context information. One of the members is bool type and this causes sparse warnings. 16:1: warning: expression using sizeof bool 60:1: warning: expression using sizeof bool 16:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) 60:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) This commit suppresses the warnings, by changing type of the member to 'unsigned int'. Additionally, this commit applies '!!' idiom to get 0/1 from 'in_interrupt()'. Fixes: a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 1622579..9c04faf 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -26,7 +26,7 @@ TRACE_EVENT(in_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -39,7 +39,7 @@ TRACE_EVENT(in_packet, __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( @@ -70,7 +70,7 @@ TRACE_EVENT(out_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -83,7 +83,7 @@ TRACE_EVENT(out_packet, __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( -- cgit v0.10.2