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 fb21b17..199408e 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 v1.1 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 5c472f3..eb85113 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 v1.1 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 009533a..df65f98 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 v1.1 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 dd7c49f..145ec4b 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 v1.1 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 145ec4b..f90139b 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 v1.1 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 e463529..76cb1b3 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 v1.1 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 97c2cac..8c7f237 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 v1.1 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 5c9caca..1cf7a32 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 v1.1 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 8f726c0..115a403 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 v1.1 From ab548d2dba63ba947287965e525cc02a15d9853d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2012 10:10:11 +0200 Subject: ALSA: hda - Fix missing Master volume for STAC9200/925x With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with vmaster hook in patch_sigmatel.c], the former Master volume control was converted to PCM. This was supposed to be covered by the vmaster control. But due to the lack of "PCM" slave definition, this didn't happen properly. The patch fixes the missing entry. Reported-by: Andrew Shadura Cc: [v3.4+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6f806d3..3d4722f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1075,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { static const char * const slave_pfxs[] = { "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "IEC958", + "Headphone", "Speaker", "IEC958", "PCM", NULL }; -- cgit v1.1 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 f3ebc38..b70964e 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 v1.1 From 1213a205f9ed27d97de3d5bed28fb085ef4853e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2012 14:58:00 +0200 Subject: ALSA: usb-audio: Fix bogus error messages for delay accounting The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fd5e982..f782ce1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / stride; int est_delay; + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; + spin_lock_irqsave(&subs->lock, flags); est_delay = snd_usb_pcm_delay(subs, runtime->rate); /* update delay with exact number of samples played */ -- cgit v1.1 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 5658bce..8d6900c 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 v1.1 From 07dc59f0988cb54fd87bd373b3b27eb2401dd811 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Sep 2012 09:39:31 +0200 Subject: ALSA: hda - Fix Oops at codec reset/reconfig snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f25c24c..1c65cc5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; @@ -2368,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; module_put(codec->owner); -- cgit v1.1 From 81cb324675eec592ab8f3038f980c074fbf7fb9b Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 11 Sep 2012 14:12:43 +0300 Subject: ALSA: compress_core: fix open flags test in snd_compr_open() O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d..eb60cb8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode), -- cgit v1.1 From c302d6133c094bda7a7ce94eac5b50c018a7ca7b Mon Sep 17 00:00:00 2001 From: Catalin Iacob Date: Sun, 9 Sep 2012 21:41:11 +0000 Subject: ALSA: hda_intel: add position_fix quirk for Asus K53E Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..228cdf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), -- cgit v1.1 From 3737e2be505d872bf2b3c1cd4151b2d2b413d7b5 Mon Sep 17 00:00:00 2001 From: Matteo Frigo Date: Wed, 12 Sep 2012 10:12:06 -0400 Subject: ALSA: ice1724: Use linear scale for AK4396 volume control. The AK4396 DAC has a linear-scale attentuator, but sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is not quite right. This patch restores the correct scale, borrowing from the ak4396 code in sound/pci/oxygen/oxygen.c. Signed-off-by: Matteo Frigo Cc: Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 764cc93d..075d5aa 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem } static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { { @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { .info = ak4396_dac_vol_info, .get = ak4396_dac_vol_get, .put = ak4396_dac_vol_put, - .tlv = { .p = db_scale_wm_dac }, + .tlv = { .p = ak4396_db_scale }, }, }; -- cgit v1.1 From 64f1e00d8edb54f5d25fb0114a46050fb8340df4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Sep 2012 15:28:56 +0200 Subject: ALSA: hda - Yet another position_fix quirk for ASUS machines ASUS X53S also suffers from the same issue as in commit c302d6133. Use POS_FIX_POSBUF for this hardware, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=47461 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 228cdf9..c4763c5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF), SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), -- cgit v1.1 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 0013afe..dc4262e 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 v1.1 From d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Sep 2012 11:57:30 +0100 Subject: ASoC: wm2000: Correct register size Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 3fd5b29..a3acb7a 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) } static const struct regmap_config wm2000_regmap = { - .reg_bits = 8, + .reg_bits = 16, .val_bits = 8, .max_register = WM2000_REG_IF_CTL, -- cgit v1.1 From 8dce30c89113e314d29d3b8f362aadff8087fccb Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 27 Sep 2012 10:26:01 +0200 Subject: ALSA: snd-usb: fix next_packet_size calls for pause case Also fix the calls to next_packet_size() for the pause case. This was missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size"). Signed-off-by: Daniel Mack Reviewed-by: Takashi Iwai Reported-and-tested-by: Christian Tefzer Cc: stable@kernel.org [ Taking directly because Takashi is on vacation - Linus ] Signed-off-by: Linus Torvalds --- sound/usb/endpoint.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d6e2bb4..060dccb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -197,7 +197,13 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep, /* no data provider, so send silence */ unsigned int offs = 0; for (i = 0; i < ctx->packets; ++i) { - int counts = ctx->packet_size[i]; + int counts; + + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + urb->iso_frame_desc[i].offset = offs * ep->stride; urb->iso_frame_desc[i].length = counts * ep->stride; offs += counts; -- cgit v1.1