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author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-05-05 17:07:06 (GMT) |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-05-05 17:07:06 (GMT) |
commit | 1c2f95480648ed7326ab2288ca0e2d35551db4be (patch) | |
tree | fa69d267423242eaad195e60c74570152e6c3d84 /sound/soc | |
parent | 59068e369b6a2a0a15b93624887525d9ec0f36e5 (diff) | |
parent | e9e7183fd2677aca24e90ca1556d4afe7436d42d (diff) | |
download | linux-fsl-qoriq-1c2f95480648ed7326ab2288ca0e2d35551db4be.tar.xz |
Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound sound fixes from Takashi Iwai:
"As good as nothing exciting here; just a few trivial fixes for various
ASoC stuff."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: omap-pcm: Free dma buffers in case of error.
ASoC: s3c2412-i2s: Fix dai registration
ASoC: wm8350: Don't use locally allocated codec struct
ASoC: tlv312aic23: unbreak resume
ASoC: bf5xx-ssm2602: Set DAI format
ASoC: core: check of_property_count_strings failure
ASoC: dt: sgtl5000.txt: Add description for 'reg' field
ASoC: wm_hubs: Make sure we don't disable differential line outputs
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/blackfin/bf5xx-ssm2602.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic23.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/wm8350.c | 11 | ||||
-rw-r--r-- | sound/soc/codecs/wm_hubs.c | 15 | ||||
-rw-r--r-- | sound/soc/omap/omap-pcm.c | 4 | ||||
-rw-r--r-- | sound/soc/samsung/s3c2412-i2s.c | 2 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 6 |
7 files changed, 27 insertions, 17 deletions
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73..b39ad35 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f9..df1e07f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c959..aa12c6b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f288..6c028c4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); - int val; + int mask, val; switch (level) { case SND_SOC_BIAS_STANDBY: @@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; + mask = 0; + + if (hubs->lineout1_se) + mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; if (hubs->lineout1_se && hubs->lineout1n_ena) val |= WM8993_LINEOUT1N_ENA; @@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, val |= WM8993_LINEOUT2P_ENA; snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, - WM8993_LINEOUT1N_ENA | - WM8993_LINEOUT1P_ENA | - WM8993_LINEOUT2N_ENA | - WM8993_LINEOUT2P_ENA, - val); + mask, val); /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd35..5a649da 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7218507..79fbeea 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1d6a80c..c88d974 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3625,10 +3625,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; |