From db8b624d55e65ad5d8211a9fef66fa7f16bd13a0 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:52 +0200 Subject: ASoC: imx-sgtl5000: fix error return code Initialize ret on the second call to imx_audmux_v2_configure_port so that the subsequent test checks that result and not the previous one. 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: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index fb21b17f17f5..199408ec4261 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { -- cgit v1.2.3 From b18e93a493626c1446f9788ebd5844d008bbf71c Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:02:53 +0200 Subject: ASoC: ux500_msp_i2s: better use devm functions and fix error return code Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap (and use resource_size for the third argument). These changes make it possible to remove the error-handling code at the end of ux500_msp_i2s_init_msp, and all of the gotos become direct returns. In the case of the second call to devm_kzalloc, the return variable ret was not initialized. Here it is changed to a direct return of -ENOMEM. A simplified version of the semantic match that finds the second 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: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 5c472f335a64..eb85113d472a 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data) { - int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; struct ux500_msp *msp; @@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, if (res == NULL) { dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } - msp->registers = ioremap(res->start, (res->end - res->start + 1)); + msp->registers = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); if (msp->registers == NULL) { dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } msp->msp_state = MSP_STATE_IDLE; @@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_err(&pdev->dev, "%s: ERROR: Failed to allocate I2S-controller!\n", __func__); - goto err_i2s_cont; + return -ENOMEM; } i2s_cont->dev.parent = &pdev->dev; i2s_cont->data = (void *)msp; @@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->i2s_cont = i2s_cont; return 0; - -err_i2s_cont: - iounmap(msp->registers); - -err_res: - devm_kfree(&pdev->dev, msp); - - return ret; } void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, @@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); device_unregister(&msp->i2s_cont->dev); - devm_kfree(&pdev->dev, msp->i2s_cont); - - iounmap(msp->registers); - - devm_kfree(&pdev->dev, msp); } MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From bc72d26bdb23c908ad52ec2d321a137d27762f08 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 19 Aug 2012 09:03:00 +0200 Subject: ASoC: am3517evm: fix error return code It was forgotten to initialize ret to the result of calling snd_soc_dai_set_sysclk, unlike at the other calls in the same 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 Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 009533ab8d18..df65f98211ec 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); -- cgit v1.2.3 From d8c3bb911f5afc32f7276c2e2e89eb58af4306ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Aug 2012 18:10:42 +0100 Subject: ASoC: dapm: Make sure we update the bias level for CODECs with no op Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) ensures that we update non-CODEC DAPM contexts but means that if a CODEC has no set_bias_level() operation it'll not be updated. Fix that. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd7c49fafd75..145ec4b56ca9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -291,6 +291,8 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); + else + dapm->bias_level = level; } else dapm->bias_level = level; -- cgit v1.2.3 From 4e872a46823c64e655d997e1e04a4b32e326aa1b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Aug 2012 18:20:49 +0100 Subject: ASoC: dapm: Don't force card bias level to be updated Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 145ec4b56ca9..f90139b5f50d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- cgit v1.2.3 From c921928661eda599d73a6a86e58bdd5aecfa18cb Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 24 Aug 2012 21:20:15 -0600 Subject: sound: tegra_alc5632: remove HP detect GPIO inversion Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren Acked-by: Andrey Danin Signed-off-by: Mark Brown Cc: # v3.4 v3.5 --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e463529b38bb..76cb1b363b71 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { -- cgit v1.2.3 From fd4fb262b31ecb06bf93defb036e72b33ddf0200 Mon Sep 17 00:00:00 2001 From: Prasad Joshi Date: Fri, 31 Aug 2012 08:55:21 +0530 Subject: ASoC: spear: correct the check for NULL dma_buffer pointer The if condition if (!buf && !buf->area) checks if the buf pointer is NULL and then dereferences it again to check if the buffer area is NULL, resulting in possible NULL dereference. Signed-off-by: Prasad Joshi Signed-off-by: Mark Brown --- sound/soc/spear/spear_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 97c2cac8e92c..8c7f23729446 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm) continue; buf = &substream->dma_buffer; - if (!buf && !buf->area) + if (!buf || !buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, -- cgit v1.2.3 From 4758be37c01c658dec5c0ad08d456fa031493de4 Mon Sep 17 00:00:00 2001 From: Heather Lomond Date: Wed, 5 Sep 2012 05:02:10 -0400 Subject: ASoC: arizona: Fix typo in 44.1kHz rates Signed-off-by: Heather Lomond Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9cacaf2d52..1cf7a32d1b21 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, -- cgit v1.2.3 From 37f45cc54cb03cac4a6b865b32bc705bb0cb1d29 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 3 Sep 2012 13:04:13 -0300 Subject: ASoC: mc13783: Remove mono support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Playing a mono track on a mc13783 codec results in incorrect playback rate. Remove mono support so that a mono track can be played correctly. Signed-off-by: Fabio Estevam Tested-by: Gaëtan Carlier Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c063f42..115a40301810 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -659,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_DAC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, @@ -670,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_CODEC, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, @@ -692,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { .id = MC13783_ID_SYNC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, -- cgit v1.2.3 From 57b2d68863f281737d8596cb3d76d89d9cc54fd8 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Sat, 1 Sep 2012 01:38:19 -0700 Subject: ASoC: samsung dma - Don't indicate support for pause/resume. The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/dma.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38c10fe..b70964ea448c 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; -- cgit v1.2.3 From a32826e4aefa905b392d2d862d51365d50d4829b Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 6 Sep 2012 17:47:33 -0600 Subject: ASoC: tegra: fix maxburst settings in dmaengine code The I2S controllers are programmed with an "attention" level of 4 DWORDs. This must match the configuration passed to the DMA driver, so that when they burst in data, they don't overflow the available FIFO space. Also, the burst size is relevant to the destination for playback, and source for capture, not vice-versa as originally written. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bcec1931..8d6900c1ee47 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; - slave_config.src_maxburst = 0; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; -- cgit v1.2.3 From 985b11fa8064d55d0d5a84e68667434598911bb2 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Fri, 14 Sep 2012 16:09:09 +0800 Subject: ASoC: wm8904: correct the index Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe48e66..dc4262eea4b7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ -- cgit v1.2.3