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2014-06-02ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixupTakashi Iwai
The conversion to a fixup table for Replacer model with ALC260 in commit 20f7d928 took the wrong widget NID for COEF setups. Namely, NID 0x1a should have been used instead of NID 0x20, which is the common node for all Realtek codecs but ALC260. Fixes: 20f7d928fa6e ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser') Cc: <stable@vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-02ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptopRonan Marquet
Correcion of wrong fixup entries add in commit ca8f0424 to replace static model quirk for PB V7900 laptop (will model). [note: the removal of ALC260_FIXUP_HP_PIN_0F chain is also needed as a part of the fix; otherwise the pin is set up wrongly as a headphone, and user-space (PulseAudio) may be wrongly trying to detect the jack state -- tiwai] Fixes: ca8f04247eaa ('ALSA: hda/realtek - Add the fixup codes for ALC260 model=will') Signed-off-by: Ronan Marquet <ronan.marquet@orange.fr> Cc: <stable@vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-30ALSA: hda/analog - Fix silent output on ASUS A8JNTakashi Iwai
ASUS A8JN with AD1986A codec seems following the normal EAPD in the normal order (0 = off, 1 = on) unlike other machines with AD1986A. Apply the workaround used for Toshiba laptop that showed the same problem. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=75041 Cc: <stable@vger.kernel.org> [3.11+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-23ALSA: hda - Fix onboard audio on Intel H97/Z97 chipsetsTakashi Iwai
The recent Intel H97/Z97 chipsets need the similar setups like other Intel chipsets for snooping, etc. Especially without snooping, the audio playback stutters or gets corrupted. This fix patch just adds the corresponding PCI ID entry with the proper flags. Reported-and-tested-by: Arthur Borsboom <arthurborsboom@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-19ALSA: pcm_dmaengine: Add check during device suspendTushar Behera
Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause() unconditinally during device suspend. In case where DMA controller doesn't support PAUSE/RESUME functionality, this call is not able to stop the DMA controller. In this scenario, audio playback doesn't resume after device resume. Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes the issue. It has been tested with audio playback on Samsung platform having PL330 DMA controller which doesn't support PAUSE/RESUME. Signed-off-by: Tushar Behera <tushar.behera@linaro.org> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-14ALSA: sb_mixer: missing return statementDan Carpenter
The if condition here was supposed to return on error but the return statement is missing. The effect is that the ->mixername is set to "???" instead of "DT019X". Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-14Merge tag 'asoc-v3.15-rc5-intel' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Intel fixes for v3.15 This is a relatively large batch of fixes for the newly added Haswell/Baytrail drivers from Intel. It's a bit larger than is good for this point in the cycle but it's all for a newly added driver so not so worrying as it might otherwise be. Some of it's integration problems, some of it's the sort of problem usually turned up in stress tests.
2014-05-14Merge tag 'asoc-v3.15-rc5-drivers' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Driver fixes for v3.15 A small set of driver fixes, nothing remarkable in itself or of any relevance outside of the driver.
2014-05-14Merge tag 'asoc-v3.15-rc5-core' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Core fixes for v3.15 A few things here: - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to have audio paths which shouldn't be present causing spurious powerups and potential audible issues for users. - Ensure the suspend->off transition doesn't have spurious transitions to prepare added to the sequence. - Fix incorrect skipping of PCM suspension for active audio streams. - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing this and Timur no longer has the boards that he was using.
2014-05-14Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linusMark Brown
2014-05-14Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', ↵Mark Brown
'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus
2014-05-13ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatileCharles Keepax
The register CLASS_D_CONTROL_1 is marked as volatile because it contains a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1 register. This causes problems for the "Speaker Switch" control, which will report an error if the CODEC is suspended because it relies on a volatile register. To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and manually keep the register cache in sync by updating both bits when changing the mute status. Reported-by: Shawn Guo <shawn.guo@linaro.org> Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Tested-by: Shawn Guo <shawn.guo@linaro.org> Signed-off-by: Mark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
2014-05-13ASoC: Intel: Fix Baytrail SST DSP firmware loadingJarkko Nikula
Commit 10df350977b1 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") caused following regression in Baytrail SST: baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with the same dma_dev device what is now used in sst_fw_new() when allocating the DMA buffer. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13ALSA: hda - mask buggy stream DMA0 for Broadwell display controllerMengdong Lin
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA postion buffer properly while DMA1 and DMA2 can work well. So this patch masks the buggy DMA0 by keeping it as opened. This is a tentative workaround, so keep the change small as Takashi suggested. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-13ALSA: hda - Add new GPU codec ID to snd-hdaAaron Plattner
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner@nvidia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-12ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params()Nicolin Chen
According to Reference Manual -- ESAI Initialization chapter, as the standard procedure of ESAI personal reset, the PCRC and PRRC registers should be remained in its reset value and then configured after T/RCCR and T/RCR configurations's done but before TE/RE's enabling. So this patch moves PCRC and PRRC settings to the end of hw_params(). Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12ASoC: fsl_esai: Only bypass sck_div for EXTAL sourceNicolin Chen
ESAI can only output EXTAL clock source directly. But for FSYS clock source, ESAI can not output it without getting through PSR PM dividers. So this patch adds an extra check in the code. Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12ASoC: fsl_esai: Fix incorrect condition within ratio range check for FPNicolin Chen
The range here from 1 to 16 is confined to FP divider only while the sck_div indicates if the calculation contains PSR and PM dividers. So for the case using PSR and PM since the sck_div is true, the range of ratio would simply become bigger than 16. So this patch fixes the condition here and adds one line comments to make the purpose here clear. Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12ASoC: dapm: Fix SUSPEND -> OFF bias sequenceLars-Peter Clausen
Currently when the DAPM context bias level is SUSPEND and the target bias level is OFF dapm_pre_sequence_async() will first transition to PREPARE and dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and then to OFF. This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE when either going to ON or away from ON. This avoids the extra unnecessary transitions. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()Lars-Peter Clausen
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm(). Also for CODEC to CODEC links the widgets are connected cross-over via a DAI link widget, meaning that the capture widget of one CODEC will be connected to the playback widget of the other and vice versa. Whereas snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of the CPU DAI to the playback widget of the CODEC DAI and the capture widget of the CPU DAI to the capture widget of the CODEC DAI. So not skipping CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create incorrect connections between the two CODECs which will cause DAPM to detect active paths where there are none and unnecessarily power up widgets. Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.") Cc: <stable@vger.kernel.org> (for 3.14+) Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPENDNicolin Chen
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-09ALSA: hda - add headset mic detect quirks for three Dell laptopsHui Wang
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255, SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292, SID: 0x10280684), the headset mic can't be detected, after apply this patch, the headset mic can work well. And on the machine with SID 0x10280684, and the Lineout and external microphone should be routed to docking, this patch also fix this problem. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-07ASoC: Intel: Fix block offset calculations.Liam Girdwood
Block offset calculations are done in the contiguous allocator so are not required here. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05ASoC: Intel: Fix check for pdata usage before dereference.Liam Girdwood
This patch fixes the following dereference check ordering. sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746) git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 0b708c87f66a15190fb43661c2320fd48c4dc6c8 vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c a4b12990 Mark Brown 2014-03-12 740 }; a4b12990 Mark Brown 2014-03-12 741 a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform) a4b12990 Mark Brown 2014-03-12 743 { a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev); a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data; 0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev; 0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0; a4b12990 Mark Brown 2014-03-12 748 a4b12990 Mark Brown 2014-03-12 @749 if (!pdata) a4b12990 Mark Brown 2014-03-12 750 return -ENODEV; a4b12990 Mark Brown 2014-03-12 751 a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05ALSA: hda - hdmi: Set converter channel count even without sinkAnssi Hannula
Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel count") channel count is no longer being set if monitor_present is 0. This is because setting the count was moved after the CA value is determined, which is only after the monitor_present check in hdmi_setup_audio_infoframe(). Unfortunately, in some cases, such as with a non-spec-compliant codec or with a problematic video driver, monitor_present is always 0. As a specific example, this seems to happen with gen1 ATV (SiI1390 codec), causing left-channel-only stereo playback (multi-channel playback has apparently never worked with this codec despite it reporting 8 channels, reason unknown). Simply setting converter channel count without setting the pin infoframe and channel mapping as well does not theoretically make much sense as this will just mean they are out-of-sync and multichannel playback will have a wrong channel mapping. However, adding back just setting the converter channel count even in no-monitor case is the safest change which at least fixes the stereo playback regression on SiI1390 codec. Do that. Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Reported-by: Stephan Raue <stephan@openelec.tv> Tested-by: Stephan Raue <stephan@openelec.tv> Cc: <stable@vger.kernel.org> # 3.12+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02ASoC: Intel: Fix stream position pointer.Liam Girdwood
Read the stream offset and presentation position from DSP memory rather than using the old estimated position. This fixes timing issues with pulseaudio. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ASoC: Intel: Fix allow hw_params to be called more than once.Liam Girdwood
hw_params() can be called multiple times. Make sure we release the DSP stream that was allocated on previous hw_params() calls before allocating a new DSP stream. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.Liam Girdwood
The Intel IOMMU requires that the ACPI device is used to allocate all DMA memory buffers. This means we need to pass the DMA device pointer into child component devices that allocate DMA memory. We also only set the DMA mask for the ACPI device now instead of for each component device. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ASoC: Intel: Fix Haswell/Broadwell DSP page table creation.Liam Girdwood
Fix page table creation on Haswell and Broadwell to remove unsafe virt_to_phys mappings and use more portable SG buffer. Use audio buffer APIs to allocate DMA buffers. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ASoC: Intel: Fix allocated block list usage when adding blocks.Liam Girdwood
Make sure we add the allocated blocks to the modules list of blocks. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ASoC: Intel: Fix block allocation so we only allocate blocks once.Liam Girdwood
Make sure we dont alloc blocks twice with requests spanning more than one block. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback dataClemens Ladisch
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus causing the PC to send the samples at a wrong rate, which results in clicks and crackles in the output. Add a workaround to detect and fix the corruption. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> [mick37@gmx.de: use sender->udh01_fb_quirk rather than ep->udh01_fb_quirk in snd_usb_handle_sync_urb()] Reported-and-tested-by: Mick <mick37@gmx.de> Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02ALSA: usb-audio: Fix deadlocks at resumingTakashi Iwai
The recent addition of the USB audio mixer suspend/resume may lead to deadlocks when the driver tries to call usb_autopm_get_interface() recursively, since the function tries to sync with the finish of the other calls. For avoiding it, introduce a flag indicating the resume operation and avoids the recursive usb_autopm_get_interface() calls during the resume. Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02ALSA: usb-audio: Save mixer status only once at suspendTakashi Iwai
The suspend callback of usb-audio driver may be called multiple times per suspend when multiple USB interfaces are bound to a single sound card instance. In such a case, it's superfluous to save the mixer values multiple times. This patch fixes it by checking the counter. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while ↵Sander Eikelenboom
DEBUG not defined This (widely used) construction: if(printk_ratelimit()) dev_dbg() Causes the ratelimiting to spam the kernel log with the "callbacks suppressed" message below, even while the dev_dbg it is supposed to rate limit wouldn't print anything because DEBUG is not defined for this device. [ 533.803964] retire_playback_urb: 852 callbacks suppressed [ 538.807930] retire_playback_urb: 852 callbacks suppressed [ 543.811897] retire_playback_urb: 852 callbacks suppressed [ 548.815745] retire_playback_urb: 852 callbacks suppressed [ 553.819826] retire_playback_urb: 852 callbacks suppressed So use dev_dbg_ratelimited() instead of this construction. Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-30ALSA: hda - add headset mic detect quirk for a Dell laptopHui Wang
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x1028067e), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-29ALSA: hda - Suppress CORBRP clear on Nvidia controller chipsTakashi Iwai
The recent commit (ca460f86521) changed the CORB RP reset procedure to follow the specification with a couple of sanity checks. Unfortunately, Nvidia controller chips seem not following this way, and spew the warning messages like: snd_hda_intel 0000:00:10.1: CORB reset timeout#1, CORBRP = 0 This patch adds the workaround for such chips. It just skips the new reset procedure for the known broken chips. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-28ALSA: hda - add headset mic detect quirk for a Dell laptopHui Wang
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x10280674), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-25ASoC: tlv320aic31xx: Convert /n to \nJoe Perches
Use a newline character appropriately. Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-24ASoC: Intel: Cancel hsw_notification_work before freeing the streamJarkko Nikula
I suppose there is a possibility that hsw_notification_work() may run after sst_hsw_stream_free() which can lead to a kernel crash since struct sst_hsw_stream is freed at that point and stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid when hsw_notification_work() is run. Reported-by: Derek Basehore <dbasehore@chromium.org> Reported-by: Wenkai Du <wenkai.du@intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-24ASoC: imx-audmux: Fix section mismatchLars-Peter Clausen
audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe() which is not marked as __init. This creates a section mismatch and a potential runtime crash (if imx_audmux_probe() is called after the .init section was dropped). This patch removes the __init annotation from audmux_debugfs_init(), which fixes the following warning: WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference from the function imx_audmux_probe() to the function .init.text:audmux_debugfs_init() Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-23ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lockKuninori Morimoto
rsnd_soc_dai_trigger() will be called after rsnd_dai_pointer_update() function which is using rsnd_lock(). Thus, it should be called from outside of rsnd_lock(). Kernel will be hangup without this patch. Special thanks to Kataoka-san Reported-by: Ryo Kataoka <ryo.kataoka.wt@renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-23ASoC: Intel: Fix audio crash due to race condition in stream deletionWenkai Du
There is a race between sst_byt_stream_free() and sst_byt_get_stream() if sst_byt_get_stream() called from sst_byt_irq_thread() context is accessing the byt->stream_list while a stream is deleted from the list. A stream is added to byt->stream_list in sst_byt_stream_new() and deleted in sst_byt_stream_free(). sst_byt_get_stream() is always protected by sst->spinlock, but the stream addition and deletion are not protected. The patch adds spinlock to both stream addition and deletion. [Jarkko: Same fix added to sst-haswell-ipc.c too] Signed-off-by: Wenkai Du <wenkai.du@intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', ↵Mark Brown
'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus
2014-04-22Merge remote-tracking branches 'asoc/fix/alc5623', 'asoc/fix/cs42l52', ↵Mark Brown
'asoc/fix/cs42l73' and 'asoc/fix/fsl-spdif' into asoc-linus
2014-04-22Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linusMark Brown
2014-04-22ASoC: jz4740: Remove Makefile entry for removed fileLars-Peter Clausen
Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") jz4740-pcm.c file, but neglected to remove the Makefile entries. Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") Reported-by: kbuild test robot <fengguang.wu@intel.com> Reported-by: Ralf Baechle <ralf@linux-mips.org> Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22ASoC: Intel: Fix audio crash due to negative address offsetWenkai Du
There were occasional ADSP crash during reboot testing: [ 11.883364] BUG: unable to handle kernel paging request at ffffc90121700000 [ 11.883380] IP: [<ffffffffc024d8bc>] sst_module_insert_fixed_block+0x24f/0x26d [snd_soc_sst_dsp] [ 11.883397] PGD 7800b067 PUD 0 [ 11.883405] Oops: 0002 [#1] SMP [ 11.886418] gsmi: Log Shutdown Reason 0x03 The virtual address, ffffc90121700000, was out of range. The virtual address is calculated by adding LPE base address with an offset: sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); The offset is calculated in sst_byt_parse_module, by subtraction of two virtual addresses dsp->addr.fw_ext and dsp->addr.lpe: block_data.offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); These virtual addresses are assigned by kernel from ioremap: sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size); In current driver code, offset is defined as unsigned int32: struct sst_module_data { ... u32 offset; /* offset in FW file */ }; Most of the time kernel assigned virtual addresses with addr.fw_ext greater than addr.lpe. But sometimes it was the other way round. Fix the problem by declaring offset as signed int32_t. Signed-off-by: Wenkai Du <wenkai.du@intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-21ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrolJarkko Nikula
Commit 9e1fda4ae158 ("ASoC: dapm: Implement mixer input auto-disable") is trying to free the widget it allocated by snd_soc_dapm_new_control() call in dapm_kcontrol_data_alloc() by adding kfree(data->widget) to dapm_kcontrol_free(). This is causing a widget double free with auto-disabled DAPM kcontrols in sound card unregistration because widgets are already freed before dapm_kcontrol_free() is called. Reason for that is all widgets are added into dapm->card->widgets list in snd_soc_dapm_new_control() and freed in dapm_free_widgets() during execution of snd_soc_dapm_free(). Now snd_soc_dapm_free() calls for different DAPM contexts happens before snd_card_free() call from where the call chain to dapm_kcontrol_free() begins: soc_cleanup_card_resources() soc_remove_dai_links() soc_remove_link_dais() snd_soc_dapm_free(&cpu_dai->dapm) soc_remove_link_components() soc_remove_platform() snd_soc_dapm_free(&platform->dapm) soc_remove_codec() snd_soc_dapm_free(&codec->dapm) snd_soc_dapm_free(&card->dapm) snd_card_free() snd_card_do_free() snd_device_free_all() snd_device_free() snd_ctl_dev_free() snd_ctl_remove() snd_ctl_free_one() dapm_kcontrol_free() This wasn't making harm with ordinary DAPM kcontrols since data->widget is NULL for them. Fixes: 9e1fda4ae158 (ASoC: dapm: Implement mixer input auto-disable) Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
2014-04-18ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume()Christian Engelmayer
Fix an incorrect sizeof() usage in sst_hsw_stream_get_volume(). sst_dsp_read() is called to read into a variable of type u32, but is passed sizeof(u32 *) for argument 'size_t bytes'. Detected by Coverity: CID 1195260. Signed-off-by: Christian Engelmayer <cengelma@gmx.at> Signed-off-by: Mark Brown <broonie@linaro.org>
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