From de050acaa1fdba4852cb195baf2bfed75368e0be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Apr 2012 20:28:10 +0100 Subject: ASoC: wm_hubs: Make sure we don't disable differential line outputs While we need to clean up unused single ended line outputs we don't want to do this if the outputs are in differential mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f2886339c..6c028c470601 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, -- cgit v1.2.3-58-ga151 From ddb6706af3cd372194cecd2cc61950519df620d7 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 24 Apr 2012 01:11:09 -0300 Subject: ASoC: dt: sgtl5000.txt: Add description for 'reg' field Add description for 'reg' field. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 2c3cd413f042..9cc44449508d 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -3,6 +3,8 @@ Required properties: - compatible : "fsl,sgtl5000". +- reg : the I2C address of the device + Example: codec: sgtl5000@0a { -- cgit v1.2.3-58-ga151 From c34ce320d9fe328e3272def20b152f39ccfa045e Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 24 Apr 2012 15:24:43 +0800 Subject: ASoC: core: check of_property_count_strings failure Signed-off-by: Richard Zhao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4e93e52b6d..b390f00b4e99 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3631,10 +3631,10 @@ 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; -- cgit v1.2.3-58-ga151 From a3a53fe1545a87337cc539f415810128bbdad465 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 25 Apr 2012 11:29:47 +0200 Subject: ASoC: bf5xx-ssm2602: Set DAI format Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC machine drivers to use the dai_links dai_fmt field to setup their DAI format. For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt, but missed to set the dai_links dai_fmt field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73f8778..b39ad356b92b 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, }, }; -- cgit v1.2.3-58-ga151 From e875c1e3e758447ba81ca450d89434b3b0496d37 Mon Sep 17 00:00:00 2001 From: Eric Bénard Date: Sun, 29 Apr 2012 17:37:57 +0200 Subject: ASoC: tlv312aic23: unbreak resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit * commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to a bug preventing resumeof the codec as regmap expects a 9 bits data register but 0xFFFF is passed in tlv320aic23_set_bias_level and this values gets cached preventing any write to the TLV320AIC23_PWR register as the final value produced by regmap is (register << 9) | value * this patch solves the problem by only working on the 9 bits the register contains. Signed-off-by: Eric Bénard Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f91a653..df1e07ffac32 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; -- cgit v1.2.3-58-ga151 From 30facd4d51d630b6cba386badd7f42456962089b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Apr 2012 20:11:55 +0100 Subject: ASoC: wm8350: Don't use locally allocated codec struct The core allocates the live copies, we shouldn't try to duplicate it and were buggy trying to do so as we were using uninitialised data for the control data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c9591ec05..aa12c6b6beeb 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 */ -- cgit v1.2.3-58-ga151 From 06412088ce98f745405b8f65cfc51ddd6b842bbf Mon Sep 17 00:00:00 2001 From: Heiko Stübner Date: Mon, 30 Apr 2012 13:17:21 +0200 Subject: ASoC: s3c2412-i2s: Fix dai registration As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai. Without this call the snd_soc_dai_ops structure isn't initialised correctly. Signed-off-by: Heiko Stuebner Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 72185078ddf8..79fbeea99d46 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) -- cgit v1.2.3-58-ga151 From fad9365bcc2a69ae16adc092e8ac192354980665 Mon Sep 17 00:00:00 2001 From: Oleg Matcovschi Date: Tue, 24 Apr 2012 19:02:02 -0700 Subject: ASoC: omap-pcm: Free dma buffers in case of error. Signed-off-by: Oleg Matcovschi Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd352d342..5a649da9122a 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; } -- cgit v1.2.3-58-ga151