From 4e38b745af76cdd93bca33258e1e33a23458f92c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 31 Oct 2012 16:07:43 +0800 Subject: ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a187830..566ea3256e2d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; -- cgit v1.2.3-58-ga151 From 8af294b472067e9034fe288d912455cc0961d1b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Feb 2013 17:48:15 +0000 Subject: ASoC: dapm: Fix handling of loops Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e1ef63d4a5c4..44a30b108683 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -488,6 +488,7 @@ struct snd_soc_dapm_path { /* status */ u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ + u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 258acadb9e7d..f3255517de79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,6 +821,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); + path->walking = 0; return widget->outputs; } } @@ -831,6 +832,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +842,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +852,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +939,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +949,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +959,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } -- cgit v1.2.3-58-ga151 From f4b828128ab64fd9dc5eec9525b38fbfeafa5c0e Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 00:23:15 +0800 Subject: ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff() 'file' is malloced in wm_adsp_load_coeff() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f8628..9af1bddc4c62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", @@ -865,7 +866,7 @@ out_fw: wm_adsp_buf_free(&buf_list); out: kfree(file); - return 0; + return ret; } int wm_adsp1_init(struct wm_adsp *adsp) -- cgit v1.2.3-58-ga151 From e8b18addee32d1f389573b4c116e67ae230216ad Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 00:35:14 +0800 Subject: ASoC: core: fix possible memory leak in snd_soc_bytes_put() 'data' is malloced in snd_soc_bytes_put() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b7e84a7cd9ee..93341deaa4b9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (params->mask) { ret = regmap_read(codec->control_data, params->base, &val); if (ret != 0) - return ret; + goto out; val &= params->mask; @@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u32 *)data)[0] |= cpu_to_be32(val); break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } } ret = regmap_raw_write(codec->control_data, params->base, data, len); +out: kfree(data); return ret; -- cgit v1.2.3-58-ga151 From b6e51600f4e983e757b1b6942becaa1ae7d82e67 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Sun, 10 Mar 2013 19:33:03 +0100 Subject: ASoC: imx-ssi: Fix occasional AC97 reset failure Signed-off-by: Sascha Hauer Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/fsl/imx-ssi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 55464a5b0706..810c7eeb7b03 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_reset) imx_ssi->ac97_reset(ac97); + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) @@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_warm_reset) imx_ssi->ac97_warm_reset(ac97); + + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } struct snd_ac97_bus_ops soc_ac97_ops = { -- cgit v1.2.3-58-ga151 From 34913fd950dc1817d466d76bdccd63443fdcbb12 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sun, 10 Mar 2013 19:33:06 +0100 Subject: ASoC: pcm030 audio fabric: remove __init from probe Remove probe function from the init section. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8e52c1485df3..eb4373840bb6 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = { .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static int __init pcm030_fabric_probe(struct platform_device *op) +static int pcm030_fabric_probe(struct platform_device *op) { struct device_node *np = op->dev.of_node; struct device_node *platform_np; -- cgit v1.2.3-58-ga151 From 7f08a89862b96d84c6dfe6c242eb010084e51d3b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Mar 2013 21:26:24 +0100 Subject: ASoC: dapm: Fix pointer dereference in is_connected_output_ep() *path is not yet initialized when we check if the widget is connected. The compiler also warns about this: sound/soc/soc-dapm.c: In function 'is_connected_output_ep': sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f3255517de79..ab621b1db105 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,7 +821,6 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); - path->walking = 0; return widget->outputs; } } -- cgit v1.2.3-58-ga151 From 4480764f57ba494e3f64003e13223c0b5ec6a2ca Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 19 Mar 2013 14:58:43 -0700 Subject: ASoC:: max98090: Remove executable bit Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches Signed-off-by: Mark Brown --- include/sound/max98090.h | 0 sound/soc/codecs/max98090.c | 0 sound/soc/codecs/max98090.h | 0 3 files changed, 0 insertions(+), 0 deletions(-) mode change 100755 => 100644 include/sound/max98090.h mode change 100755 => 100644 sound/soc/codecs/max98090.c mode change 100755 => 100644 sound/soc/codecs/max98090.h (limited to 'sound/soc') diff --git a/include/sound/max98090.h b/include/sound/max98090.h old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h old mode 100755 new mode 100644 -- cgit v1.2.3-58-ga151 From 59d9cc2a5073ab4b8c8f8bdbacf230a538abc55d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 18 Mar 2013 18:57:23 +0100 Subject: ASoC: spear_pcm: Update to new pcm_new() API Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/spear/spear_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5a1148..5e7aebe1e664 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) -- cgit v1.2.3-58-ga151 From f7ba716f1e704a00d682a8697108f9c86497c551 Mon Sep 17 00:00:00 2001 From: Silviu-Mihai Popescu Date: Sat, 16 Mar 2013 13:45:34 +0200 Subject: ASoC: core: fix invalid free of devm_ allocated data The objects allocated by devm_* APIs are managed by devres and are freed when the device is detached. Hence there is no need to use kfree() explicitly. Signed-off-by: Silviu-Mihai Popescu Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93341deaa4b9..507d251916af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4199,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4208,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } } -- cgit v1.2.3-58-ga151 From 417a1178f1bf3cdc606376b3ded3a22489fbb3eb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 15 Mar 2013 11:26:15 +0100 Subject: ASoC: dma-sh7760: Fix compile error The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/sh/dma-sh7760.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8fc4fdd..1a8b03e4b41b 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; -- cgit v1.2.3-58-ga151 From 0eaa6cca1f75e12e4f5ec62cbe887330fe3b5fe9 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 26 Mar 2013 14:41:05 +0900 Subject: ASoC: core: Fix to check return value of snd_soc_update_bits_locked() It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is success. So just check return value is negative. Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 507d251916af..ff4b45a5d796 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val = val << shift; ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); - if (ret != 0) + if (ret < 0) return ret; if (snd_soc_volsw_is_stereo(mc)) { -- cgit v1.2.3-58-ga151 From fa40ef208c955bfe21f53913f51f297ac3237e95 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Mar 2013 16:39:01 +0000 Subject: ASoC: compress: Cancel delayed power down if needed When a new stream is being opened it is necessary to cancel any delayed power down of the audio. [Fixed unused variable -- broonie] Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b5b3db71e253..ed0bfb0ddb96 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } -- cgit v1.2.3-58-ga151 From a9b977ecd3dbc5d4f0fe0b3d5c66d284859b1f2a Mon Sep 17 00:00:00 2001 From: Prathyush K Date: Tue, 2 Apr 2013 16:53:01 +0530 Subject: ASoC: Samsung: return error if drvdata is not set This patch fixes a possible crash in case drvdata for the secondary device is not set. Signed-off-by: Prathyush K Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e336a7c..f1fc06419560 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1107,6 +1107,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (samsung_dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); + if (!sec_dai) { + dev_err(&pdev->dev, "Unable to get drvdata\n"); + return -EFAULT; + } snd_soc_register_dai(&sec_dai->pdev->dev, &sec_dai->i2s_dai_drv); asoc_dma_platform_register(&pdev->dev); -- cgit v1.2.3-58-ga151 From c6f9b1eb0e5df468891eff17f981b76c86f95f3a Mon Sep 17 00:00:00 2001 From: Prathyush K Date: Tue, 2 Apr 2013 16:53:02 +0530 Subject: ASoC: Samsung: set drvdata before adding secondary device Currently, a new platform device is created for secondary device by calling platform_device_register_resndata and then the drvdata is set for this device. The following patch has been added to driver core: "driver core: fix possible missing of device probe". This results in the added device getting probed immediately but the drvdata for the secondary device is not yet set. This patch removes the platform_device_register_resndata call and instead calls platform_device_alloc, platform_set_drvdata and platform_device_add which fixes the above issue. Signed-off-by: Prathyush K Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f1fc06419560..6bbeb0bf1a73 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; + int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) @@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; + dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ - i2s->pdev = platform_device_register_resndata(NULL, - "samsung-i2s-sec", -1, NULL, 0, NULL, 0); + i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); if (IS_ERR(i2s->pdev)) return NULL; - } - /* Pre-assign snd_soc_dai_set_drvdata */ - dev_set_drvdata(&i2s->pdev->dev, i2s); + platform_set_drvdata(i2s->pdev, i2s); + ret = platform_device_add(i2s->pdev); + if (ret < 0) + return NULL; + } return i2s; } -- cgit v1.2.3-58-ga151 From 5aa995e83ac7727b7705431e6eb2b317c59b95ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 3 Apr 2013 11:00:01 +0200 Subject: ASoC: tegra: Don't claim to support PCM pause and resume The tegra dmaengine driver does not support pausing and resuming a DMA stream. The tegra PCM driver still claims to support pause and resume though and implements them by stopping and restarting the stream. This is not what an application using pause/resume would expect. Usually applications have support for working around PCMs which do not support suspend and resume, so don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and use the default snd_dmaengine_pcm_trigger callback. Signed-off-by: Lars-Peter Clausen Reviewed-by: Stephen Warren Tested-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c925ab0adeb6..5e2c55c5b255 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -43,8 +43,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels_min = 2, @@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int tegra_pcm_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: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_START); - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - default: - return -EINVAL; - } - return 0; -} - static int tegra_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = tegra_pcm_hw_params, .hw_free = tegra_pcm_hw_free, - .trigger = tegra_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = snd_dmaengine_pcm_pointer, .mmap = tegra_pcm_mmap, }; -- cgit v1.2.3-58-ga151 From f1ca493b0b5e8f42d3b2dc8877860db2983f47b6 Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Tue, 9 Apr 2013 17:13:59 +0200 Subject: ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running The Charge Pump needs the DSP clock to work properly, without it the bypass to HP/LINEOUT is not working properly. This requirement is not mentioned in the datasheet but has been confirmed by Mark Brown from Wolfson. Signed-off-by: Alban Bedel Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8903.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 134e41c870b9..f8a31ad0b203 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + { "Charge Pump", NULL, "CLK_DSP" }, + { "Left Headphone Output PGA", NULL, "Charge Pump" }, { "Right Headphone Output PGA", NULL, "Charge Pump" }, { "Left Line Output PGA", NULL, "Charge Pump" }, -- cgit v1.2.3-58-ga151 From f6f629f8332ea70255f6c60c904270640a21a114 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Apr 2013 13:19:26 +0100 Subject: ASoC: wm5102: Correct lookup of arizona struct in SYSCLK event Reported-by: Ryo Tsutsui Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5102.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b82bbf584146..34d0201d6a78 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -584,7 +584,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = codec->control_data; const struct reg_default *patch = NULL; int i, patch_size; -- cgit v1.2.3-58-ga151