From d0db84e713eaaccea2a435e1625fb3150b335f4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Aug 2012 15:37:47 +0300 Subject: ASoC: omap-mcbsp: Fix 6pin mux configuration The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org (3.4+) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8..d33c48b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { -- cgit v0.10.2 From 48a08bab3066a9452216f8c52e0d6f35566de04d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 8 Aug 2012 00:47:21 -0300 Subject: ASoC: mxs: Fix the name of the SoC family SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f..b6fa776 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help -- cgit v0.10.2 From 0865a75d4166bddc533fd50831829ceefb94f9b0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Aug 2012 16:51:34 -0300 Subject: ASoC: imx-ssi: Remove mono support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier Tested-by: Gaëtan Carlier Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c..81d7728 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- cgit v0.10.2 From ed36081350d2ca4f692f04c6a2d24d1e3a339da1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:12:52 +0200 Subject: ALSA: hda - Add codec->pcm_format_first flag Introduced a new flag to set up the PCM stream format at first before the stream_id and channel tag. Some codecs (e.g. CA0132) seem preferring this over stream_id -> format order. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..598b9e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1386,6 +1386,44 @@ int snd_hda_codec_configure(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_codec_configure); +/* update the stream-id if changed */ +static void update_pcm_stream_id(struct hda_codec *codec, + struct hda_cvt_setup *p, hda_nid_t nid, + u32 stream_tag, int channel_id) +{ + unsigned int oldval, newval; + + if (p->stream_tag != stream_tag || p->channel_id != channel_id) { + oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); + newval = (stream_tag << 4) | channel_id; + if (oldval != newval) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + newval); + p->stream_tag = stream_tag; + p->channel_id = channel_id; + } +} + +/* update the format-id if changed */ +static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p, + hda_nid_t nid, int format) +{ + unsigned int oldval; + + if (p->format_id != format) { + oldval = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_STREAM_FORMAT, 0); + if (oldval != format) { + msleep(1); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, + format); + } + p->format_id = format; + } +} + /** * snd_hda_codec_setup_stream - set up the codec for streaming * @codec: the CODEC to set up @@ -1400,7 +1438,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, { struct hda_codec *c; struct hda_cvt_setup *p; - unsigned int oldval, newval; int type; int i; @@ -1413,29 +1450,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p = get_hda_cvt_setup(codec, nid); if (!p) return; - /* update the stream-id if changed */ - if (p->stream_tag != stream_tag || p->channel_id != channel_id) { - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - p->stream_tag = stream_tag; - p->channel_id = channel_id; - } - /* update the format-id if changed */ - if (p->format_id != format) { - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, 0); - if (oldval != format) { - msleep(1); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - p->format_id = format; - } + + if (codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); + if (!codec->pcm_format_first) + update_pcm_format(codec, p, nid, format); + p->active = 1; p->dirty = 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..7fbc1bc 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,6 +861,7 @@ struct hda_codec { unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ + unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ -- cgit v0.10.2 From 55cf87fe0e9783e25f442be1e48b8319d86131ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:15:55 +0200 Subject: ALSA: hda - Fix superfluous "-in" suffix from CA0132 capture items Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index d0d3540..2685590 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -988,12 +988,12 @@ static void ca0132_config(struct hda_codec *codec) /* Mic-in */ spec->input_pins[0] = 0x12; - spec->input_labels[0] = "Mic-In"; + spec->input_labels[0] = "Mic"; spec->adcs[0] = 0x07; /* Line-In */ spec->input_pins[1] = 0x11; - spec->input_labels[1] = "Line-In"; + spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; } -- cgit v0.10.2 From 27ebeb0b1b5bb26908e485a7e8bd2ec30366ffef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:20:18 +0200 Subject: ALSA: hda - Use the standard PCM ops for CA0132 Now with the workaround using codec->pcm_format_first flag, we can clean up the home-baked codes in patch_ca0132.c. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2685590..31512a0 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -464,50 +464,17 @@ exit: } /* - * PCM stuffs + * PCM callbacks */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) +static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd("ca0132_setup_stream: " - "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + struct ca0132_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); } -/* - * PCM callbacks - */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -515,10 +482,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); } static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, @@ -526,92 +491,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dacs[0]); - - return 0; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } /* * Digital out */ -static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format); - - return 0; -} - -static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_out); - - return 0; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -/* - * Analog capture - */ -static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); } -static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->adcs[substream->number]); - - return 0; -} - -/* - * Digital capture - */ -static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo, +static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format); - - return 0; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); } -static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct ca0132_spec *spec = codec->spec; - - ca0132_cleanup_stream(codec, spec->dig_in); - - return 0; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } /* @@ -621,6 +539,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_playback_pcm_open, .prepare = ca0132_playback_pcm_prepare, .cleanup = ca0132_playback_pcm_cleanup }, @@ -630,10 +549,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup - }, }; static struct hda_pcm_stream ca0132_pcm_digital_playback = { @@ -641,6 +556,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = { .channels_min = 2, .channels_max = 2, .ops = { + .open = ca0132_dig_playback_pcm_open, + .close = ca0132_dig_playback_pcm_close, .prepare = ca0132_dig_playback_pcm_prepare, .cleanup = ca0132_dig_playback_pcm_cleanup }, @@ -650,10 +567,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .ops = { - .prepare = ca0132_dig_capture_pcm_prepare, - .cleanup = ca0132_dig_capture_pcm_cleanup - }, }; static int ca0132_build_pcms(struct hda_codec *codec) @@ -961,6 +874,9 @@ static void ca0132_config(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* line-outs */ cfg->line_outs = 1; cfg->line_out_pins[0] = 0x0b; /* front */ -- cgit v0.10.2 From 8e13fc1c5f694a6ae4032c7f94103c137136733f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Aug 2012 17:26:54 +0200 Subject: ALSA: hda - Add missing SPDIF I/O setup for CA0132 CA0132 driver had some codes to handle the S/PDIF I/O, but the actual setups of pins and converters were missing. Now the pins are added. Also, fixed a few points triggering invalid codec verbs and mixer elements since the digital I/O audio widgets on CA0132 have no amp. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 31512a0..9c0ec0a 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); } - if (dac) + if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); } @@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } - if (adc) + if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); } @@ -841,18 +841,16 @@ static int ca0132_build_controls(struct hda_codec *codec) spec->dig_out); if (err < 0) return err; - err = add_out_volume(codec, spec->dig_out, "IEC958"); + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); if (err < 0) return err; + /* spec->multiout.share_spdif = 1; */ } if (spec->dig_in) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); if (err < 0) return err; - err = add_in_volume(codec, spec->dig_in, "IEC958"); - if (err < 0) - return err; } return 0; } @@ -912,6 +910,16 @@ static void ca0132_config(struct hda_codec *codec) spec->input_labels[1] = "Line"; spec->adcs[1] = 0x08; spec->num_inputs = 2; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static void ca0132_init_chip(struct hda_codec *codec) -- cgit v0.10.2 From 0d624275720a4b01217693eb80d967a0d5f1f3a3 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 8 Aug 2012 20:40:31 +0530 Subject: ASoC: Davinci: McASP: Flush the FIFO before enabling FIFO should be flushed before it is enabled for the first time. This fixes the I/O errors reported by the ASoC core on a fresh boot Signed-off-by: Vaibhav Bedia Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bf..ce5e5cd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* enable FIFO */ + if (dev->txnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } mcasp_start_tx(dev); } else { - if (dev->rxnumevt) /* enable FIFO */ + if (dev->rxnumevt) { /* enable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } mcasp_start_rx(dev); } } -- cgit v0.10.2 From 8b5eae137b91cb2db15fe2c5a913cafde4629339 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Thu, 9 Aug 2012 18:08:40 -0400 Subject: ASoC: bfin: fix memory leak in sport3 controller driver Signed-off-by: Scott Jiang Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index 318c5ba5..dfb7443 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create); void sport_delete(struct sport_device *sport) { + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); sport_free_resource(sport); + kfree(sport); } EXPORT_SYMBOL(sport_delete); -- cgit v0.10.2 From 52c0eee3329b08dfd912a59e0246e21026308301 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jul 2012 18:23:35 +0100 Subject: ASoC: wm8962: Don't duplicate bias level management in resume The core will bring the bias level up for us since we use idle_bias_off, duplicating this may be harmful. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa9ce9d..ce67200 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3733,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev) regcache_sync(wm8962->regmap); - regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - - /* VMID back to 2x250k for standby */ - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK, 0x100); - return 0; } -- cgit v0.10.2 From 15676937e6a7e98d37f4c1eaa0e7b3c111627fce Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Thu, 9 Aug 2012 10:10:54 +0100 Subject: ASoC: wm8994: Add missing dapm routes for WM8958 rev A Signed-off-by: Chris Rattray Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef031..6c9eeca 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4038,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, -- cgit v0.10.2 From d34e4e00adbbc91ff9fc96ed9a4e4b65161868da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 15:47:15 +0200 Subject: ALSA: platform: Check CONFIG_PM_SLEEP instead of CONFIG_PM When CONFIG_PM is set but CONFIG_PM_SLEEP is unset, SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to compile warnings. For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM. Reported-by: Arnd Bergmann Signed-off-by: Takashi Iwai diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 0d7b25e..4e1fda7 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = { .prepare = pxa2xx_ac97_pcm_prepare, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pxa2xx_ac97_do_suspend(struct snd_card *card) { @@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = { .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .pm = &pxa2xx_ac97_pm_ops, #endif }, diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index eb4ceb7..98554f4 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -534,7 +534,7 @@ out_put_pclk: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_abdac_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index bf47025..3c8d3ba7 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1134,7 +1134,7 @@ err_snd_card_new: return retval; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atmel_ac97c_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1128b35..5a34355 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int loopback_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index f7d3bfc..54bb664 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_dummy_suspend(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 6ca59fc..ef17129 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) pcspkr_stop_sound(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int pcsp_suspend(struct device *dev) { struct snd_pcsp *chip = dev_get_drvdata(dev); @@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL); #define PCSP_PM_OPS &pcsp_pm #else #define PCSP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index f5ceb6f..210cafe 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_pmac_driver_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); -- cgit v0.10.2 From 144dad99ef6ad10c8c8ebe787d08157c4a94201f Mon Sep 17 00:00:00 2001 From: James Ralston Date: Thu, 9 Aug 2012 09:38:59 -0700 Subject: ALSA: hda_intel: Add Device IDs for Intel Lynx Point-LP PCH This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH Signed-off-by: James Ralston Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..60882c6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, LPT_LP}," "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + /* Lynx Point-LP */ + { PCI_DEVICE(0x8086, 0x9c21), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v0.10.2 From fb099cb712e878b9eb4e78dd6b268312a0b2b50f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 9 Aug 2012 18:44:37 +0100 Subject: ASoC: core: Upgrade the severity of probe deferral errors to dev_err() In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..c501af6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } -- cgit v0.10.2 From 9c30959884261518de46e0b79ac7a902e6ddd147 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 23 Jun 2012 11:25:43 +0100 Subject: MAINTAINERS: Add entries for Wolfson Arizona class devices Signed-off-by: Mark Brown diff --git a/MAINTAINERS b/MAINTAINERS index 94b823f..dba8941 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7665,23 +7665,28 @@ S: Supported F: Documentation/hwmon/wm83?? F: arch/arm/mach-s3c64xx/mach-crag6410* F: drivers/clk/clk-wm83*.c +F: drivers/extcon/extcon-arizona.c F: drivers/leds/leds-wm83*.c F: drivers/gpio/gpio-*wm*.c +F: drivers/gpio/gpio-arizona.c F: drivers/hwmon/wm83??-hwmon.c F: drivers/input/misc/wm831x-on.c F: drivers/input/touchscreen/wm831x-ts.c F: drivers/input/touchscreen/wm97*.c -F: drivers/mfd/wm8*.c +F: drivers/mfd/arizona* +F: drivers/mfd/wm*.c F: drivers/power/wm83*.c F: drivers/rtc/rtc-wm83*.c F: drivers/regulator/wm8*.c F: drivers/video/backlight/wm83*_bl.c F: drivers/watchdog/wm83*_wdt.c +F: include/linux/mfd/arizona/ F: include/linux/mfd/wm831x/ F: include/linux/mfd/wm8350/ F: include/linux/mfd/wm8400* F: include/linux/wm97xx.h F: include/sound/wm????.h +F: sound/soc/codecs/arizona.? F: sound/soc/codecs/wm* WORKQUEUE -- cgit v0.10.2 From de64c0ee7dbcbfbbe63bd9ea45783d87babc6452 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 10 Aug 2012 12:22:58 +0300 Subject: ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read() This function returns its own error codes instead of normal negative error codes. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ff..a71d1c1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1); -- cgit v0.10.2 From 14bc9c6dc694e2d7930802f7afd275de25ef8394 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 10 Aug 2012 13:29:32 +0200 Subject: ALSA: hda - Fix panned "Beep Playback Switch" When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315..d26ae65 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && !beep->enabled) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = 0; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v0.10.2 From e037cb4a54e26b5f55f856e0e7445cfcfb2f3d31 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 10 Aug 2012 14:11:58 +0200 Subject: ALSA : hda - bug fix on checking the supported power states of a codec The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..629131a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- cgit v0.10.2 From 61f5d61ef94d7082d96494e2a6dd79de2b4437d2 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Wed, 8 Aug 2012 11:34:43 +0530 Subject: ASoC: Samsung: Fix build error MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes the following build error: In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from sound/soc/samsung/pcm.c:23: arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: error: redefinition of ‘struct s3c2410_dma_client’ arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Signed-off-by: Sachin Kamat Signed-off-by: Sachin Kamat Acked-by: Kukjin Kim Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index b7b2a1f..89b0646 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,7 @@ #include #include -#include +#include #include "dma.h" #include "pcm.h" -- cgit v0.10.2 From 088c820b732dbfd515fc66d459d5f5777f79b406 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 13 Aug 2012 14:11:10 +0800 Subject: ALSA: hda - fix Copyright debug message As spec said, 1 indicates no copyright is asserted. Signed-off-by: Wang Xingchao Cc: Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258..6894ec6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_EMPHASIS) snd_iprintf(buffer, " Preemphasis"); if (digi1 & AC_DIG1_COPYRIGHT) - snd_iprintf(buffer, " Copyright"); + snd_iprintf(buffer, " Non-Copyright"); if (digi1 & AC_DIG1_NONAUDIO) snd_iprintf(buffer, " Non-Audio"); if (digi1 & AC_DIG1_PROFESSIONAL) -- cgit v0.10.2 From 14ebd8a8c15e9fed638120bdb93f1a814e13d6a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:12 +0100 Subject: ASoC: wm5102: Add missing input PGA routes Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16..496ce9a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -639,6 +639,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v0.10.2 From 17c3f7e8f3ef796a9db3b22f3797188d0e7ac88c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Aug 2012 15:40:22 +0100 Subject: ASoC: wm5110: Add missing input PGA routes Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f70..01ebbcc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "IN4L PGA", NULL, "IN4L" }, + { "IN4R PGA", NULL, "IN4R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- cgit v0.10.2 From 12022a785328fdf61a3e1a4bc34db0098dabe839 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Aug 2012 16:28:36 +0100 Subject: ASoC: jack: Always notify full jack status Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7..0c17293 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); -- cgit v0.10.2 From 265d931a9e9a7e290faa5e2145f4b2ebf38ea84c Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 13 Aug 2012 17:10:46 +0200 Subject: ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch Some Conexant devices (e g CX20590) have no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d26ae65..0849aac 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -231,15 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); +static bool ctl_has_mute(struct snd_kcontrol *kcontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return query_amp_caps(codec, get_amp_nid(kcontrol), + get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE; +} + /* get/put callbacks for beep mute mixer switches */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep && !beep->enabled) { + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = 0; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -263,6 +270,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, enable |= *valp; snd_hda_enable_beep_device(codec, enable); } + if (!ctl_has_mute(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- cgit v0.10.2 From 3bdcff70b6cd049e6f4437b955850f5db83653cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 17:42:11 +0200 Subject: ALSA: lx6464es: Add a missing error check Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44541 Signed-off-by: Takashi Iwai diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index d1ab437..5579b08 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip) /* hardcoded device name & channel count */ err = snd_pcm_new(chip->card, (char *)card_name, 0, 1, 1, &pcm); + if (err < 0) + return err; pcm->private_data = chip; -- cgit v0.10.2 From e9ba389c5ffc4dd29dfe17e00e48877302111135 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Aug 2012 12:32:00 +0200 Subject: ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream A PCM capture stream on usb-audio causes a scheduling-while-atomic BUG, as reported in the bugzilla entry below. It's because snd_usb_endpoint_start() is called at first at trigger START for a capture stream, and this function contains the left-over EP deactivation codes. The problem doesn't happen for a playback stream because the function is called at PCM prepare time, which can sleep. This patch fixes the BUG by moving the EP deactivation code into the PCM prepare callback. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 Cc: [v3.5+] Signed-off-by: Takashi Iwai diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0f647d2..c411812 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - /* just to be sure */ - deactivate_urbs(ep, 0, 1); - wait_clear_urbs(ep); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a1298f3..62ec808 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; + /* clear the pending deactivation on the target EPs */ + deactivate_endpoints(subs); + /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) -- cgit v0.10.2 From 5e68fb3cab23b327e9f15803607e697d7eea1966 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Aug 2012 14:11:09 +0200 Subject: ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040cc..ea5775a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- cgit v0.10.2 From 939d5044b117302cabdd30833685d9f214e9bff6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 13:08:23 +0100 Subject: ASoC: wm5102: Remove DRC2 It will be removed from future device revisions. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 496ce9a..e33d327 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -684,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), -- cgit v0.10.2 From ccf795847a38235ee4a56a24129ce75147d6ba8f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 22:36:04 +0100 Subject: ASoC: wm9712: Fix microphone source selection Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz Signed-off-by: Mark Brown Cc: stable@vger.kernel.org diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f16fb36..fd74b88 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, -- cgit v0.10.2 From 28c42c28309244d0b15d1b385e33429d59997679 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:37:28 +0100 Subject: ASoC: wm9712: Fix inverted capture volume The capture volume increases with the register value so it shouldn't be flagged as inverted. Reported-by: Christoph Fritz Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd74b88..c6d2076 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), -SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), +SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), -- cgit v0.10.2 From 94f3ec6b2222eb5c0af0c784f0656ff5b909d870 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 18 Aug 2012 18:55:15 +0300 Subject: sound: oss/sb_audio: prevent divide by zero bug Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014..b2b3c01 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst; -- cgit v0.10.2 From 8513915accc611e576dbebb93422c257e7e68be8 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sat, 18 Aug 2012 17:43:05 -0700 Subject: ALSA: fix pcm.h kernel-doc warning and notation Fix kernel-doc warning in and add function name to make the kernel-doc notation complete. Warning(include/sound/pcm.h:1081): No description found for parameter 'substream' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c75c0d1..cdca2ab 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1075,7 +1075,8 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); /** - * Get a string naming the direction of a stream + * snd_pcm_stream_str - Get a string naming the direction of a stream + * @substream: the pcm substream instance */ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) { -- cgit v0.10.2 From aaf265c22e48f10c94ad04d23b6ab0c88f554d35 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:58 +0200 Subject: ALSA: sound/atmel/abdac.c: fix error return code Initialize retval before returning from a failed call to ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 98554f4..277ebce 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) dac->regs = ioremap(regs->start, resource_size(regs)); if (!dac->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto out_free_card; } -- cgit v0.10.2 From 0c23e46eb4878422c25351ff54ab0fe80c643809 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:57 +0200 Subject: ALSA: sound/atmel/ac97c.c: fix error return code In the first case, the second test of whether retval is negative is redundant. It is dropped and the previous and subsequent tests are combined. In the second case, add an initialization of retval on failure of ioremap. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 3c8d3ba7..9052aff 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (cpu_is_at32ap7000()) { - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); - } + if (cpu_is_at32ap7000() && retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (!chip->regs) { dev_dbg(&pdev->dev, "could not remap register memory\n"); + retval = -ENOMEM; goto err_ioremap; } -- cgit v0.10.2 From 4d8ce1c9966663bad69e738952179f3cc52710bf Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:56 +0200 Subject: ALSA: sound/pci/ctxfi/ctatc.c: fix error return code Initialize err before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262..2f6e9c7 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_connect_resources(atc); atc->timer = ct_timer_new(atc); - if (!atc->timer) + if (!atc->timer) { + err = -ENOMEM; goto error1; + } err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) -- cgit v0.10.2 From ae970eb45d8a1ea4506be23c3f776225b9575d0e Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:55 +0200 Subject: ALSA: sound/pci/sis7019.c: fix error return code Initialize rc before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434e..805ab6e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, - sis)) { + rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, + sis); + if (rc) { dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; } -- cgit v0.10.2 From b17cbdd85f84c8323189416da6e9701d2793b0e5 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:54 +0200 Subject: ALSA: sound/pci/rme9652/hdspm.c: fix error return code Convert a nonnegative error return code to a negative one, as returned elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8ac871..b12308b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, snd_printk(KERN_ERR "HDSPM: " "unable to kmalloc Mixer memory of %d Bytes\n", (int)sizeof(struct hdspm_mixer)); - return err; + return -ENOMEM; } hdspm->port_names_in = NULL; -- cgit v0.10.2 From c86b93628e5649fd7bb0574b570a51b2b02d586c Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:59 +0200 Subject: ALSA: sound/ppc/snd_ps3.c: fix error return code Initialize ret before returning on failure, as done elsewhere in the function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 1aa52ef..9b18b52 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) GFP_KERNEL); if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); + ret = -ENOMEM; goto clean_preallocate; } pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, -- cgit v0.10.2 From c41999a23929f30808bae6009d8065052d4d73fd Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Aug 2012 11:17:00 +0200 Subject: ALSA: hda - don't create dysfunctional mixer controls for ca0132 It's possible that these amps are settable somehow, e g through secret codec verbs, but for now, don't create the controls (as they won't be working anyway, and cause errors in amixer). Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/1038651 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9c0ec0a..49750a9 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) { + snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) { + snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid); + return 0; + } sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } -- cgit v0.10.2 From 535b6c51fe8293c88ce919cdfc4390c67a1cb6d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 21:25:22 +0200 Subject: ALSA: hda - Fix leftover codec->power_transition When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: [v3.5+] Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c3077d5..f560051 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4454,6 +4454,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; } -- cgit v0.10.2