May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Broxton Machine
by Vinod Koul 10 May '16

10 May '16

Broxton uses SSP5 for audio on Intel board, so add the 6 SSPs and then machine driver which uses RT298 codec Pardha Saradhi K (1): ASoC: Intel: Skylake: Add more SSP DAIs Ramesh Babu (1): ASoC: Intel: Add Broxton-P machine driver sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_rt298.c | 353 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 72 ++++++++ 4 files changed, 442 insertions(+) create mode 100644 sound/soc/intel/boards/bxt_rt298.c -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: da7213: clocking related improvements for codec
by Adam Thomson 10 May '16

10 May '16

This is a collection of updates with regards to clocking on da7213 codec, including power efficiency & stability improvements. Patches are based on v4.6-rc7 kernel. Adam Thomson (5): ASoC: da7213: Add DAI DAPM event to control DAI clocks ASoC: da7213: Add checking of SRM lock status before enabling DAI ASoC: da7213: Default PC counter to free-running when DAI disabled ASoC: da7213: Update PLL ranges to improve locking at frequency boundary ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk sound/soc/codecs/da7213.c | 99 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 35 ++++++++++------- 2 files changed, 102 insertions(+), 32 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH 0/7] ALSA: drop reuse of incoming packet parameter for outgoing packet parameter
by Takashi Sakamoto 10 May '16

10 May '16

Hi, This patchset updates my former RFC, and go for merging to Linux 4.7. [alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html The aim of this patchset is to remove a feature from firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. The feature is based on our understanding of IEC 61883-6. In the specification, a function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect corresponding drivers to satisfy the feature. On the other hand, the feature brings inconveniences to this module. When this feature is enabled, a software IRQ context processes packets for both of IR/IT contexts. This consumes more time over local CPUs than handling one context. Furthermore, cycle count for outgoing packets becomes improper because they're for incoming packets. For these reasons, this patchset removes the feature from ALSA firewire stack. Changes: - The last patch is newly added. Regards Takashi Sakamoto (7): ALSA: bebob: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reuse of incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software interrupt context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag sound/firewire/amdtp-stream.c | 98 +++++++++------------------ sound/firewire/amdtp-stream.h | 36 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 101 +++++++--------------------- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------ sound/firewire/oxfw/oxfw-stream.c | 3 +- sound/firewire/tascam/tascam-stream.c | 26 ++++--- 9 files changed, 94 insertions(+), 258 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: pcm: Bail out when chmap is already present
by Takashi Iwai 10 May '16

10 May '16

When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a chmap has been already assigned, it returns as an error due to the conflicting snd_ctl_add() result. However, this also clears the already assigned chmap_kctl field via pcm_chmap_ctl_private_free(), and becomes inconsistent in the later operation. This patch adds the check of the conflicting chmap kctl before actually trying to allocate / assign. The check failure is treated as a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is basically a driver bug and having the stack trace would help developers to figure out the bad code path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0aca39762ed0..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; -- 2.8.2

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[alsa-devel] [PATCH 3/4] ASoC: ab8500-codec: remove platform data based probe
by Arnd Bergmann 10 May '16

10 May '16

The ux500 platform hasn't used board files for a long time, and nothing defines a ab8500_codec_platform_data, so we can just remove the probing based on that and always use device tree properties directly. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/ab8500-codec.c | 33 ++++----------------------------- 1 file changed, 4 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 8b1d0c1a7839..2fc89155f14a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2464,45 +2464,20 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) struct device *dev = codec->dev; struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); - struct ab8500_platform_data *pdata; + struct ab8500_codec_platform_data codec_pdata; struct filter_control *fc; int status; dev_dbg(dev, "%s: Enter.\n", __func__); - /* Setup AB8500 according to board-settings */ - pdata = dev_get_platdata(dev->parent); + ab8500_codec_of_probe(dev, np, &codec_pdata); - if (np) { - if (!pdata) - pdata = devm_kzalloc(dev, - sizeof(struct ab8500_platform_data), - GFP_KERNEL); - - if (pdata && !pdata->codec) - pdata->codec - = devm_kzalloc(dev, - sizeof(struct ab8500_codec_platform_data), - GFP_KERNEL); - - if (!(pdata && pdata->codec)) - return -ENOMEM; - - ab8500_codec_of_probe(dev, np, pdata->codec); - - } else { - if (!(pdata && pdata->codec)) { - dev_err(dev, "No codec platform data or DT found\n"); - return -EINVAL; - } - } - - status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + status = ab8500_audio_setup_mics(codec, &codec_pdata.amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); return status; } - status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + status = ab8500_audio_set_ear_cmv(codec, codec_pdata.ear_cmv); if (status < 0) { pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", __func__, status); -- 2.7.0

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[alsa-devel] bug: reconfig broken on kernel 4.5
by Jochen Henneberg 10 May '16

10 May '16

Hi, it looks as if 'reconfig' feature is broken since the sysfs attributes have been moved to device attributes. In hda_sysfs.c:reconfig_codec() the device is cleared with snd_hda_codec_reset() which calls snd_hdac_device_unregister() which removes the sysfs entries (including the currently in-use 'reconfig'), the re-registration fails afterwards in snd_hda_codec_build_controls(). I am not sure when things got broken, but snd_hda_codec_reset() called from reconfig_codec() unregisters the device and removes the sysfs entries where in the past it only stripped all configurations (pcms, controls ...) from the device. Regards, Jochen

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[alsa-devel] [PATCH] ALSA: Fix subwoofer pin on ASUS N751 and N551
by Takashi Iwai 10 May '16

10 May '16

From: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Subwoofer does not work out of the box on ASUS N751/N551 laptops. This patch fixes it. Patch tested on N751 laptop. N551 part is not tested, but according to [1] and [2] this laptop requires similar changes, so I included them in the patch. 1. https://github.com/honsiorovskyi/asus-n551-hda-fix 2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781 Signed-off-by: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b5cd19876fc..4918ffa5ba68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6426,6 +6426,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6672,6 +6673,15 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6700,6 +6710,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.8.2

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[alsa-devel] Kernel config issue: SND_HDA_RECONFIG depends on SND_HDA_HWDEP
by Jochen Henneberg 09 May '16

09 May '16

Hi, with kernel 4.5 (and it looks as if this has not changed since then) the SND_HDA_RECONFIG sysfs entries still show up in the hwdep devices. Enabling SND_HDA_RECONFIG without SND_HDA_HWDEP is worth nothing, but the dependency has been removed from Kconfig. The commits from 13aeaf68 / 'Manage each codec instance individually' read as if the sysfs entries should show up at the codec driver but they do not. Regards, Jochen

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[alsa-devel] [PATCH 0/5] Fix poll error returns
by Charles Keepax 09 May '16

09 May '16

We return negative values from the file_operations poll callback in a few places, this callback returns an unsigned int and is expected to only return the poll specific flags. This series fixes up these issues and makes a couple of small tidy ups to the code around the edges of those functions. I noticed this issue whilst doing some additional testing on my propagation of compressed stream error series, but I decided to push these up separately as it is worth getting the fixes in now and not tying them up with that patch chain which is taking longer to get merged. Also I included the first patch of that chain (Replace complex if statement with switch) because it is a trivial tidy up and might as well get merged now as well. Thanks, Charles Charles Keepax (5): ALSA: pcm: Fix poll error return codes ALSA: compress: Use snd_compr_get_poll on error path ALSA: compress: Remove pointless NULL check ALSA: compress: Fix poll error return codes ALSA: compress: Replace complex if statement with switch sound/core/compress_offload.c | 25 ++++++++++++------------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 14 insertions(+), 15 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/3 v3] firewire-lib: support tracepoints to dump a part of packet data
by Takashi Sakamoto 09 May '16

09 May '16

Hi, This patchset updates my RFCv2, and go for merging. [alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html The motivation of this patchset is to check the sequence of packet from/to audio and music units on IEEE 1394 bus. When the units have some quirks or ALSA firewire stack includes some bugs, we can work to seek them with Linux tracing framework. Changes: - Improve commit comments. - Improve Python3 script in commit comment for NODATA/NOINFO packets. - Move the position of trace_out_acket() so that the index of packet starts 0. Takashi Sakamoto (3): ALSA: firewire-lib: compute the value of second field in cycle count for IT context ALSA: firewire-lib: compute the value of second field in cycle count for IR context ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 98 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 77 ++++++++++++++++++++++------- 3 files changed, 160 insertions(+), 18 deletions(-) create mode 100644 sound/firewire/amdtp-stream-trace.h -- 2.7.4

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[alsa-devel] [PATCH alsa-lib] conf/ucm: ROCKCHIP-I2S: add Rockchip I2S UCM config.
by Enric Balletbo i Serra 09 May '16

09 May '16

Taken from the ChromeOS sources, this configuration was tested on Veyron Jerry based Chromebook from Google. Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/ROCKCHIP-I2S/HiFi.conf | 94 +++++++++++++++++++++++++++++ src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf | 6 ++ 3 files changed, 101 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/ROCKCHIP-I2S/HiFi.conf create mode 100644 src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index e6a6325..d1d51e5 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 ROCKCHIP-I2S diff --git a/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf new file mode 100644 index 0000000..7bfe995 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf @@ -0,0 +1,94 @@ +SectionVerb { + Value { + OutputDspName "speaker_eq" + MinBufferLevel "512" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 20" + cset "name='Headset Mic Switch' off" + cset "name='Int Mic Switch' on" + + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + ] +} + +SectionDevice."Headphone".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + OutputDspName "" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Speaker Switch' off" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + ] + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Speaker Switch' on" + ] +} + +SectionDevice."Mic".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Headset Mic Switch' on" + cset "name='Record Path DC Blocking' on" + ] + + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headset Mic Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] +} diff --git a/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf new file mode 100644 index 0000000..0c399b0 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf @@ -0,0 +1,6 @@ +Comment "Rockchip card" + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.1.0

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[alsa-devel] [PATCH] pcm: softvol: fix conversion of TLVs min_db and max_dB value
by Jörg Krause 09 May '16

09 May '16

Both, min_dB and max_dB, are floating type whereas the TLV is (always) unsigned. The problem with the conversion of a negative floating-point number into an unsigned integer is, that the behavior is undefined. This may, depending on the platform, result in a wrong TLV, i.e. for the default values of min_dB (-51dB) and max_dB (0dB), alsactl generates the following state on an ARM cpu build with GCC: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } With the fix applied, alsactl stores the correct TLV: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Replaces: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107733.html Fixes: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107628.html Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..061080d 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -658,8 +658,8 @@ static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) unsigned int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); - tlv[2] = svol->min_dB * 100; - tlv[3] = (svol->max_dB - svol->min_dB) * 100 / svol->max_val; + tlv[2] = (int)(svol->min_dB * 100); + tlv[3] = (int)((svol->max_dB - svol->min_dB) * 100 / svol->max_val); return snd_ctl_elem_tlv_write(svol->ctl, &cinfo->id, tlv); } -- 2.8.2

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[alsa-devel] [PATCH] ASoC: pcm: allow changing the playback/capture rates for symmetric links
by Peter Rosin 09 May '16

09 May '16

The below program fails on a dai link with symmetric rates without this patch. The patch makes it work. #include <sys/soundcard.h> #include <unistd.h> #include <fcntl.h> #include <stropts.h> #include <stdio.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22050; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4fa86..1e876ff23524 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -237,6 +237,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate, channels, sample_bits, symmetry, i; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (!cpu_dai->capture_active) + return 0; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!cpu_dai->playback_active) + return 0; + } + rate = params_rate(params); channels = params_channels(params); sample_bits = snd_pcm_format_physical_width(params_format(params)); -- 2.1.4

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[alsa-devel] [PATCH 0/2] wm5110: fixes to ANC clock control
by Richard Fitzgerald 09 May '16

09 May '16

The register for ANC clock control has a combination of volatile and non-readback bits and these two patches fix the way we write to this register. Both patches must be taken together to ensure correct operation. Richard Fitzgerald (2): mfd: wm5110: ARIZONA_CLOCK_CONTROL should be volatile ASoC: arizona: No need to update_bits when writing AEC clock control drivers/mfd/wm5110-tables.c | 1 + sound/soc/codecs/arizona.c | 3 +-- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 09 May '16

09 May '16

From: Shreyas NC <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can be changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v3: add bound check for firmware file sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 118 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 ++ sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 5 ++ 6 files changed, 245 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..a3cd5d7ab246 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,118 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + uuid_le *uuid_mod; + + tbl = ctx->tbl; + num = ctx->num_modules; + uuid_mod = (uuid_le *)uuid; + + for (i = 0; i < num; i++) { + if (uuid_le_cmp(*uuid_mod, tbl[i].uuid) == 0) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int snd_skl_parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + uuid_le *uuid_bin; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Read the UUID(GUID) from FW Manifest. + * The 16 byte UUID is of the format: + * XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX + * Populate the UUID table to store module_id + * and loadable flags for the module. + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + + /* + * we check if current pointer is larger than file size from + * base value to check excceding the file while parsing + */ + if ((const char *)mod_entry >= buf + ctx->fw->size) { + dev_err(ctx->dev, + "Exceeds file bound: Entry %d Ptr %p\n", + i, mod_entry); + + return -EIO; + } + + uuid_bin = (uuid_le *)mod_entry->uuid.id; + memcpy(&tbl[i].uuid, uuid_bin, sizeof(tbl[i].uuid)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_skl_parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..cfdb24c45035 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,107 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u8 id[16]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + uuid_le uuid; + int module_id; + int is_loadable; +}; + +int snd_skl_parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index bec4a7c486fd..df7964fd09c3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -25,6 +25,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -70,7 +71,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -84,6 +85,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = snd_skl_parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a1f4478fabcb..f690b016203c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1566,6 +1567,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, w->priv = mconfig; memcpy(&mconfig->guid, &dfw_config->uuid, 16); + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; -- 1.9.1

2 2

[alsa-devel] [PATCH v2 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 09 May '16

09 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. History: v2: Fix memory leak on realloc failure. The original obj pointer is reserved and so it can be freed at last by tplg_elem_free(). Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 26 +++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 133 insertions(+), 58 deletions(-) -- 2.5.0

2 7

[alsa-devel] [PATCH v2 1/2] ALSA: hda: fix the missing ptr initialization
by Vinod Koul 09 May '16

09 May '16

ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- 1.9.1

2 4

[alsa-devel] [PATCH] ALSA: dice: add support for M-Audio Profire 610 and perhaps 2626
by Takashi Sakamoto 09 May '16

09 May '16

M-Audio Profire 610 has an unexpected value in version field of its config ROM, thus ALSA dice driver is not assigned to the model due to a mismatch of modalias. This commit adds an entry to support the model. I expect the entry is also for Profire 2626. I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a part of Extended Application Protocol (EAP). $ cd linux-firewire-utils/src $ ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ------------------------------------------------------------ 400 04047689 bus_info_length 4, crc_length 4, crc 30345 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000d6c04 company_id 000d6c | 410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002 root directory ------------------------------------------------------------ 414 000695fe directory_length 6, crc 38398 418 03000d6c vendor 41c 8100000a --> descriptor leaf at 444 420 17000011 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ------------------------------------------------------------ 430 0004fb14 directory_length 4, crc 64276 434 12000d6c specifier id 438 130100d1 version 43c 17000011 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ------------------------------------------------------------ 444 0004b8e4 leaf_length 4, crc 47332 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4175 "M-Au" 454 64696f00 "dio" descriptor leaf at 458 ------------------------------------------------------------ 458 00053128 leaf_length 5, crc 12584 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 50726f46 "ProF" 468 69726520 "ire " 46c 36313000 "610" descriptor leaf at 470 ------------------------------------------------------------ 470 00053128 leaf_length 5, crc 12584 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 50726f46 "ProF" 480 69726520 "ire " 484 36313000 "610" $ cat /proc/asound/card1/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc0:000100000000 notification: 00000040 nick name: FW610 clock select: internal 48000 enable: 1 status: locked 48000 ext status: 00000040 sample rate: 48000 version: 1.0.4.0 clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\ ... Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96fe68f4..25e9f77 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -317,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); -- 2.7.4

2 4

[alsa-devel] [PATCH] pcm: softvol: fix signedness of tlv data type
by Jörg Krause 08 May '16

08 May '16

For the softvol plugin the TLV data are written wrong, e.g. for the default values of min_dB = -51 dB and max_dB = 0 dB, alsactl generates the following state: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } As both min_dB and max_dB can be negative numbers, the tlv type must not be unsigned. This fixes the TLV and the correct state is generated: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..1748667 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -655,7 +655,7 @@ static void snd_pcm_softvol_dump(snd_pcm_t *pcm, snd_output_t *out) static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) { - unsigned int tlv[4]; + int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); tlv[2] = svol->min_dB * 100; @@ -768,7 +768,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, } } else if (svol->max_val > 1) { /* check TLV availability */ - unsigned int tlv[4]; + int tlv[4]; err = snd_ctl_elem_tlv_read(svol->ctl, &cinfo->id, tlv, sizeof(tlv)); if (err < 0) add_tlv_info(svol, cinfo); -- 2.8.2

2 2

[alsa-devel] [PATCH 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 08 May '16

08 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 24 ++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 131 insertions(+), 58 deletions(-) -- 2.5.0

5 11

Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Kailang 08 May '16

08 May '16

Hi Takashi, Could you receive this mail? Thanks. BR, Kailang ________________________________ From: Kailang Sent: Wednesday, May 04, 2016 4:13 PM To: Takashi Iwai (tiwai(a)suse.de) Cc: (alsa-devel(a)alsa-project.org) Subject: New codecs support for ALC234/ALC274/ALC294 Hi Takashi, We have new codecs for ALC234/ALC274/ALC294. Attach file is the support patch. BR, Kailang

2 1

Re: [alsa-devel] ALSA: hda - Fix white noise on Asus UX501VW headset
by Takashi Iwai 08 May '16

08 May '16

On Thu, 05 May 2016 16:08:02 +0200, Kaho Ng wrote: > > For reducing the noise from the headset output on ASUS UX501VW, > call the existing fixup, alc_fixup_headset_mode_alc668(), additionally. > > Thread: https://bbs.archlinux.org/viewtopic.php?id=209554 > > Signed-off-by: Kaho Ng <ngkaho1234(a)gmail.com> > --- > diff -ruNp a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > --- a/sound/pci/hda/patch_realtek.c 2016-04-20 14:44:02.000000000 +0800 > +++ b/sound/pci/hda/patch_realtek.c 2016-05-03 16:31:30.815677146 +0800 > @@ -6424,6 +6424,7 @@ enum { > ALC668_FIXUP_DELL_DISABLE_AAMIX, > ALC668_FIXUP_DELL_XPS13, > ALC662_FIXUP_ASUS_Nx50, > + ALC668_FIXUP_ASUS_UX501VW, > }; > > static const struct hda_fixup alc662_fixups[] = { > @@ -6670,6 +6671,10 @@ static const struct hda_fixup alc662_fix > .chained = true, > .chain_id = ALC662_FIXUP_BASS_1A > }, > + [ALC668_FIXUP_ASUS_UX501VW] = { > + .type = HDA_FIXUP_FUNC, > + .v.func = alc_fixup_headset_mode_alc668, > + }, > }; There is already a fixup entry (ALC668_FIXUP_HEADSET_MODE) doing the very same thing, so you don't have to redefine it. > static const struct snd_pci_quirk alc662_fixup_tbl[] = { > @@ -6692,6 +6697,7 @@ static const struct snd_pci_quirk alc662 > SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), > + SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_ASUS_UX501VW), And just add the quirk entry like SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), Could you resubmit the fix patch after adjusting like this? thanks, Takashi

1 0

Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Takashi Iwai 08 May '16

08 May '16

On Wed, 04 May 2016 10:12:41 +0200, Kailang wrote: > > Hi Takashi, > > We have new codecs for ALC234/ALC274/ALC294. > Attach file is the support patch. Thanks, applied now. Takashi

1 0

[alsa-devel] [patch] ALSA: isa/wavefront: prevent some out of bound writes
by Dan Carpenter 08 May '16

08 May '16

"header->number" can be up to USHRT_MAX and it comes from the ioctl so it needs to be capped. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff..718d5e3 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics

2 1

Re: [alsa-devel] [PATCH] infoleak fix1 in timer
by Takashi Iwai 08 May '16

08 May '16

On Tue, 03 May 2016 22:44:07 +0200, Kangjie Lu wrote: > > The stack object “tread” has a total size of 32 bytes. Its field > “event” and “val” both contain 4 bytes padding. These 8 bytes > padding bytes are sent to user without being initialized. > > Signed-off-by: Kangjie Lu <kjlu(a)gatech.edu> Thanks, I applied now all three patches. But I rephrased the subjects to more meaningful ones. Better to give a bit more informative subject line at the next time. Takashi

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