October 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Audio miniconference schedule
by Mark Brown 24 Oct '18

24 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh: https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda: - Virtualization (Liam) - SoundWire (Liam) - Byte controls (Vinod) - GObject introspection for language bindings (Sakamoto-san) - Integrating the ALSA control core (Sakamoto-san) - AGL (Stephane/Ronan) - Y2038 (broonie) - AVB (Liam) - DSPs (Liam) - Sound Open Firmware What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event. It also seems good to mention that there's another audio event going on around ELC: https://blogs.gnome.org/uraeus/2018/09/24/getting-the-team-together-to-revo… which is a hackathon for a new userspace stack called PipeWire - if we have someone attending who knows about that I think it'd be good to talk about it too.

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[alsa-devel] [PATCH - Intervals v2 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 24 Oct '18

24 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) simular to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- > On 10/18/18 12:57, Takashi Iwai wrote: > This change looks risky. The snd_interval_value() might be called > even if the interval isn't reduced to a single value. Rather check > openmin instead. If I would do "if (i->openmin)" on an interval of (x x+1) x+1 would be returned. But this would result in the second issue which I have tried to describe in the commit message: x has to be returned otherwise it is not guaranteed that buffer_size >= 2x period_size. And this would result in continuously Xruns. Therefore I would like to prefer "if (i->openmin && !i->openmax)" diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..d9a30b2 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (i->openmin && !i->openmax) + return i->max; return i->min; } -- 2.7.4

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Re: [alsa-devel] GFP_ATOMIC in Intel code
by Takashi Iwai 24 Oct '18

24 Oct '18

On Tue, 23 Oct 2018 10:49:27 +0200, Vinod wrote: > > On 22-10-18, 12:22, Pierre-Louis Bossart wrote: > > Looking at the latest for-next branch, by accident I saw a number of > > GFP_ATOMIC references. > > > > With my extended break I don't remember much but didn't we agree that none > > of those were needed? If yes I'll send a patch to fix all this. > > > I thought that someone had fixed up all these instances, Looking back at the git history, "someone" was mostly you :) A few of them below seem to be new codes that copied the bad past examples, some are overlooked one, and some are mailbox codes that aren't 100% sure to safe to replace (although I guess it is). > looks like few > missed, so yes please do fix them up and Welcome back :-) I hope you enjoyed the time-off, Pierre. thanks, Takashi > > > Thanks > > > > -Pierre > > > > > > /data/pbossart/ktest/broonie-next/sound/soc/intel$ git grep ATOMIC > > atom/sst/sst_pvt.c: msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); > > atom/sst/sst_pvt.c: msg->mailbox_data = > > kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); > > boards/cht_bsw_rt5672.c: drv = devm_kzalloc(&pdev->dev, sizeof(*drv), > > GFP_ATOMIC); > > boards/glk_rt5682_max98357a.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/kbl_da7219_max98927.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > boards/skl_hda_dsp_generic.c: ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), > > GFP_ATOMIC); > > -- > ~Vinod >

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[alsa-devel] [PATCH - AAF PCM plugin 0/7] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 23 Oct '18

23 Oct '18

Hi all, The AAF plugin RFC series v2 has been around on alsa-devel ML for more than two weeks so I'm moving forward and sending it as PATCH series now. The first 5 patches from this series are pretty much the same from the previous version. The news are the top 2 patches which add more functionality to the plugin. Patch 6/7 implements the dump() callback which has been useful for debugging purposes and Patch 7/7 adds support for direct read/write transfers. For general context information about the AAF plugin, I'm quoting below the cover letter from the RFC series v1: "This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit / receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener." For further information about what has been previously discussed, please refer to RFC series v1 and v2 archives in [5] and [6]. Finally, all versions of this series can be also found in my alsa-plugins tree in github [7]. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.html [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… [7] https://github.com/aguedes/alsa-plugins Andre Guedes (7): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support aaf: Implement dump() ioplug callback aaf: Add support for direct read/write transfers Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1360 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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[alsa-devel] [GIT PULL] ASoC fixes for the next release
by Mark Brown 23 Oct '18

23 Oct '18

The following changes since commit 7f91e2af1a4a2c34fc2e8fb046c722e1a9c85399: ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks (2018-10-21 16:46:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.0-2 for you to fetch changes up to 6be0f96d799f487f05eb412d362d5a1747d665c2: ASoC: stm32: sai: fix master clock naming (2018-10-23 10:51:45 +0100) ---------------------------------------------------------------- ASoC: Additional fixes for the next release A couple of fixes for build bot issues in the STM32 SAI driver. ---------------------------------------------------------------- Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-)

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[alsa-devel] Segfault in pulseaudio since Alsa 1.1.7
by Hyperion 23 Oct '18

23 Oct '18

Hi, Â Since the upgrade to Alsa 1.1.7, I get a segfault at pulseaudio startup. This have been discussed and documented here : [1]https://www.linuxquestions.org/questions/slackware-14/current-fri-oc t-19-puseaudio-segfauts-4175640773/ Â It seems that I found the origin of the bug : reverting these 2 commits fixes the problem : [2]http://git.alsa-project.org/?p=alsa-...792dc809c908cc [3]http://git.alsa-project.org/?p=alsa-...5dc6e1594291e7 Then no more segfault in pulseaudio when accessing the USB soundcard. Â Please could you fix it ? Â Regards jp References Visible links 1. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu… 2. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=fc3f5d51062f… 3. http://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff;h=6b996865126d… Hidden links: 4. https://www.linuxquestions.org/questions/slackware-14/current-fri-oct-19-pu…

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[alsa-devel] [alsa-plugins-1.17] Regression in JACK plugin
by tet 23 Oct '18

23 Oct '18

Hi, after updating to alsa-plugins-1.17 ALSA throws: "ALSA lib pcm_plug.c:1273:(_snd_pcm_plug_open) Unknown field playback_ports My ~/.asourdrc is working in 1.16 and it does not look like the syntax changed for the plugin when looking at the readme. pcm.jack { type plug slave.pcm { type jack capture_ports { 0 system:capture_1 1 system:capture_1 } playback_ports { 0 system:playback_1 1 system:playback_2 } } hint.description "JACK Audio Kit" } pcm.!default { @func getenv vars [ PCM ] default jack } Any ideas?

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[alsa-devel] [RFC PATCH 0/4] Initial prototype of DAPM sample rate domains
by Charles Keepax 23 Oct '18

23 Oct '18

Here is a first pass at adding dapm sample rate domain support. Things are still pretty rough in quite a few places and I expect some major refactoring before they are ready to merge, but things are far enough along for people to look at the approach I am taking to the problem. And should provide enough for discussion at the mini-conference, although hopefully I will get some more done next week as well. The basic high level summary is two new concepts are added to ASoC, one being a sample rate domain and the other being a domain group. The domain groups represent widgets that are limited by hardware restrictions to always run at the same sample rate as each other. Sample rate domains represent a slot on a device for a sample rate. Both of these are currently limited to be within the scope of a single component. The core will track which groups are connected together, then as a group is powered up it will be allocated a domain based on the other groups it is connected to. Domain groups are implemented as a new type of DAPM widget and the domains themselves are a new thing which devices add through their component drivers. As we have previously discussed my intention would be that we also add bridges between domains although I haven't started work on that yet. The current code is limited to within a single component and I think that the bridges will allow better support for propagating rates between components as well as obviously for SRCs. Currently the SRCs on wm5110 are just being handled as points where the domain groups don't connect to another peer. This allows them to function but doesn't accurately convey things like the limitations which input rates support which output rates on the ISRC. Thanks, Charles Charles Keepax (4): ASoC: dapm: Add support for hw_free on CODEC to CODEC links ASoC: dapm: Add support for a rate domain widget ASoC: domain: Add sample rate domain support ASoC: arizona: Add rate domain support include/sound/soc-dapm.h | 14 ++ include/sound/soc-domain.h | 98 +++++++++++ include/sound/soc.h | 8 + sound/soc/Makefile | 2 +- sound/soc/codecs/arizona.c | 255 ++++++++++++++++++++++------ sound/soc/codecs/arizona.h | 76 ++++++++- sound/soc/codecs/wm5110.c | 414 ++++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 8 + sound/soc/soc-dapm.c | 57 ++++++- sound/soc/soc-domain.c | 412 ++++++++++++++++++++++++++++++++++++++++++++ 10 files changed, 1183 insertions(+), 161 deletions(-) create mode 100644 include/sound/soc-domain.h create mode 100644 sound/soc/soc-domain.c -- 2.11.0

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[alsa-devel] [asoc:for-4.20 333/424] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 23 Oct '18

23 Oct '18

Hi Rakesh, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 1c8bc7b3de5e76cb89aacdc7be1475a028af505f commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/424] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/2] ASoC: stm32: sai: fix kbuild errors
by Olivier Moysan 23 Oct '18

23 Oct '18

These patches fix following compilation errors: - >> sound/soc/stm/stm32_sai_sub.c:260:16: error: field 'hw' has incomplete type - >> sound/soc/stm/stm32_sai_sub.c:412:3: warning: 'strncat' specified bound 6 equals source length [-Wstringop-overflow=] Olivier Moysan (2): ASoC: stm32: add clock dependency for sai ASoC: stm32: sai: fix master clock naming sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_sai_sub.c | 5 ++--- 2 files changed, 3 insertions(+), 3 deletions(-) -- 2.7.4

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