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 042b92c185cbd7b4291710255510ae76b2d7797b Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 22 Aug 2012 16:10:43 +0200 Subject: ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakers This fixes an issue with a machine where there were no speakers, but GPIO0 had to be data=1 for the headphone to be functioning. I'm not sure if we need a more advanced patch to solve all possible cases, but if so, this patch would still provide a minor optimisation. BugLink: https://bugs.launchpad.net/bugs/1040077 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea5775a1a7db..3edd73c3d361 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4543,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (cfg->speaker_outs == 0) + return; + for (i = 0; i < cfg->line_outs; i++) { if (presence) break; -- 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 983f6b93818aa62fbc74c37fcb8a482718a19252 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 09:18:01 -0700 Subject: ALSA: hda - Avoid unnecessary parameter read for EPSS EPSS parameter should be static, so we can read it once and remember. This also allows more easily to override the wrong EPSS capability reported from a codec by changing the flag in the codec initialization step. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++++++-- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f560051a949e..f25c24c743f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->epss = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_EPSS); + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3543,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ - bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); - msleep(epss ? 10 : 100); + msleep(codec->epss ? 10 : 100); } /* repeat power states setting at most 10 times*/ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7fbc1bcaf1a9..e5a7e19a8071 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -862,6 +862,7 @@ struct hda_codec { unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ + unsigned int epss:1; /* supporting EPSS? */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ -- cgit v1.2.3 From c36b5b054aaf14d68261970e3769398110e636d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 09:20:13 -0700 Subject: ALSA: hda - Don't trust codec EPSS bit for IDT 92HD83xx & co These codecs seem reporting EPSS but require longer delay for the proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set correctly even after D3. In this patch, codec->epss flag is overridden for avoid the misbehavior. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3edd73c3d361..6f806d3e56bb 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5534,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); } + codec->epss = 0; /* longer delay needed for D3 */ codec->no_trigger_sense = 1; codec->spec = spec; -- 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 015618b902ae8e28705b7af9b4668615fea48ddd Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 29 Aug 2012 13:17:05 +0200 Subject: ALSA: snd-usb: Fix URB cancellation at stream start Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by: Daniel Mack Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 11 +++++++++-- sound/usb/endpoint.h | 2 +- sound/usb/pcm.c | 13 +++++-------- 3 files changed, 15 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c41181202688..b896c5559524 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -799,7 +799,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, /** * snd_usb_endpoint_start: start an snd_usb_endpoint * - * @ep: the endpoint to start + * @ep: the endpoint to start + * @can_sleep: flag indicating whether the operation is executed in + * non-atomic context * * A call to this function will increment the use count of the endpoint. * In case it is not already running, the URBs for this endpoint will be @@ -809,7 +811,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, * * Returns an error if the URB submission failed, 0 in all other cases. */ -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) { int err; unsigned int i; @@ -821,6 +823,11 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; + /* just to be sure */ + deactivate_urbs(ep, 0, can_sleep); + if (can_sleep) + wait_clear_urbs(ep); + ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index ee2723fb174f..a8e60c1408e5 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 62ec808ed792..1546577ae458 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } -static int start_endpoints(struct snd_usb_substream *subs) +static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) { int err; @@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs) snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); ep->data_subs = subs; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); return err; @@ -239,7 +239,7 @@ static int start_endpoints(struct snd_usb_substream *subs) snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); return err; @@ -544,13 +544,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; - /* clear the pending deactivation on the target EPs */ - deactivate_endpoints(subs); - /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) - return start_endpoints(subs); + return start_endpoints(subs, 1); return 0; } @@ -1175,7 +1172,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream switch (cmd) { case SNDRV_PCM_TRIGGER_START: - err = start_endpoints(subs); + err = start_endpoints(subs, 0); if (err < 0) return err; -- cgit v1.2.3 From 03d2f44e967b3c2cf79a6dfb904c8880616c7f83 Mon Sep 17 00:00:00 2001 From: Pavel Roskin Date: Thu, 30 Aug 2012 17:11:17 -0400 Subject: ALSA: snd-usb: use list_for_each_safe for endpoint resources snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- sound/usb/card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index d5b5c3388e28..4a469f0cb6d4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, struct snd_usb_audio *chip) { struct snd_card *card; - struct list_head *p; + struct list_head *p, *n; if (chip == (void *)-1L) return; @@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_stream_disconnect(p); } /* release the endpoint resources */ - list_for_each(p, &chip->ep_list) { + list_for_each_safe(p, n, &chip->ep_list) { snd_usb_endpoint_free(p); } /* release the midi resources */ -- cgit v1.2.3 From fbcfbf5f673847657ccd98afb4d8e13af7fdc372 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 30 Aug 2012 18:52:29 +0200 Subject: ALSA: snd-usb: restore delay information Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB frame counter") were unfortunately lost during the refactoring of the snd-usb driver in 3.5. This patch adds them back, restoring the correct delay information behaviour. Signed-off-by: Daniel Mack Cc: Pierre-Louis Bossart Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 29 ++++++++++++++++++++++++++--- 1 file changed, 26 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1546577ae458..5ceb8f1d63fb 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1091,7 +1091,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -1109,12 +1118,26 @@ static void retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + + /* + * Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms. + */ + if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", + est_delay, subs->last_delay); + spin_unlock_irqrestore(&subs->lock, flags); } -- cgit v1.2.3 From 245baf983cc39524cce39c24d01b276e6e653c9e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 30 Aug 2012 18:52:30 +0200 Subject: ALSA: snd-usb: fix calls to next_packet_size In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by: Daniel Mack Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 13 +------------ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 7 ++++++- 3 files changed, 8 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b896c5559524..d6e2bb49c59c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) * * For implicit feedback, next_packet_size() is unused. */ -static int next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, ep->retire_data_urb(ep->data_subs, urb); } -static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep, - struct snd_urb_ctx *ctx) -{ - int i; - - for (i = 0; i < ctx->packets; ++i) - ctx->packet_size[i] = next_packet_size(ep); -} - /* * Prepare a PLAYBACK urb for submission to the bus. */ @@ -370,7 +361,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; } - prepare_outbound_urb_sizes(ep, ctx); prepare_outbound_urb(ep, ctx); } else { retire_inbound_urb(ep, ctx); @@ -857,7 +847,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) goto __error; if (usb_pipeout(ep->pipe)) { - prepare_outbound_urb_sizes(ep, urb->context); prepare_outbound_urb(ep, urb->context); } else { prepare_inbound_urb(ep, urb->context); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index a8e60c1408e5..cbbbdf226d66 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -21,6 +21,7 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, struct snd_usb_endpoint *sender, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5ceb8f1d63fb..e80b6687f43a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1029,6 +1029,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, struct urb *urb) { struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_usb_endpoint *ep = subs->data_endpoint; struct snd_urb_ctx *ctx = urb->context; unsigned int counts, frames, bytes; int i, stride, period_elapsed = 0; @@ -1040,7 +1041,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { - counts = ctx->packet_size[i]; + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + /* set up descriptor */ urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; -- cgit v1.2.3 From 2e4a263ca80a203ac6109f5932722a716c265395 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 30 Aug 2012 18:52:31 +0200 Subject: ALSA: snd-usb: fix cross-interface streaming devices Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface") saved us some unnecessary calls to snd_usb_set_interface() but ignored the fact that there is at least one device out there which operates on two endpoint in different interfaces simultaniously. Take care for this by catching the case where data and sync endpoints are located on different interfaces and calling snd_usb_set_interface() between the start of the two endpoints. Signed-off-by: Daniel Mack Reported-by: Robert M. Albrecht Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e80b6687f43a..fd5e982fc98c 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -236,6 +236,21 @@ static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx); + if (err < 0) { + snd_printk(KERN_ERR + "%d:%d:%d: cannot set interface (%d)\n", + subs->dev->devnum, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx, err); + return -EIO; + } + } + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; -- 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 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 6f806d3e56bb..3d4722f0a1ca 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 v1.2.3 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 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 fd5e982fc98c..f782ce19bf5a 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 v1.2.3 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 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 f25c24c743f9..1c65cc5e3a31 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 v1.2.3 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 ec2118d0e27a..eb60cb8dbb8a 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 v1.2.3 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 60882c62f180..228cdf93fa29 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 v1.2.3 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 764cc93dbca4..075d5aa1fee0 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 v1.2.3 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 228cdf93fa29..c4763c52eaf6 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 v1.2.3 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