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

[alsa-devel] [RESEND][PATCH] ASoC: Intel: Power down links before turning off display audio power
by Sriram Periyasamy 01 Aug '18

01 Aug '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++------- sound/soc/intel/skylake/skl.c | 11 ++++++----- 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 0cdceca2b216..c98c4c2babc9 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2235,12 +2235,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); @@ -2249,7 +2243,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(ebus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 670ff9aaca55..c62e474717ff 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -762,6 +762,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -771,11 +777,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); -- 2.7.4

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[alsa-devel] [PATCH v5 0/8] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 01 Aug '18

01 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v5 is based on Takashi's topic/drm_audio_component merged on top on Marks' for-next branch - the merge is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (6): ASoC: Intel: Skylake: extend widget handling ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 487 ++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/intel/Kconfig | 19 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 130 ++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 176 ++++++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 +++- sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 100 ++++- sound/soc/intel/skylake/skl.h | 12 +- 18 files changed, 1111 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.14.1

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[alsa-devel] [PATCH v4 0/4] Add support for audio on SDM845 SoC
by Rohit kumar 01 Aug '18

01 Aug '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v3: - Added Module_license in common.c - Merged apq8096.c machine driver change with common api patch - Addressed comments by Srinivas, Mark and Vinod. Rohit kumar (4): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: dt-bindings: Update dt binding name for apq8096 ASoC: qcom: Add support to parse common audio device nodes ASoC: qcom: add sdm845 sound card support .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 ++++++ sound/soc/qcom/Kconfig | 8 + sound/soc/qcom/Makefile | 4 +- sound/soc/qcom/apq8096.c | 111 +------- sound/soc/qcom/common.c | 112 ++++++++ sound/soc/qcom/common.h | 12 + sound/soc/qcom/sdm845.c | 286 +++++++++++++++++++++ 8 files changed, 525 insertions(+), 103 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ASoC: soc-pcm: Use delay set in pointer function
by Akshu Agrawal 01 Aug '18

01 Aug '18

There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c Also, in some cases cpu dai used is generic and the pcm driver needs to set delay. This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/soc-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 98be04b..b1a2bc2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1179,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1203,7 +1206,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) } delay += codec_delay; - runtime->delay = delay; + runtime->delay += delay; return offset; } -- 1.9.1

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[alsa-devel] [asoc:for-4.19 199/204] sound/soc//meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'; did you mean 'clk_set_rate'?
by kbuild test robot 01 Aug '18

01 Aug '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: m68k-allyesconfig (attached as .config) compiler: m68k-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 c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=m68k All errors (new ones prefixed by >>): sound/soc//meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc//meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'; did you mean 'clk_set_rate'? [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ clk_set_rate cc1: some warnings being treated as errors vim +225 sound/soc//meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] 答复: A bug about cache inconsistency report
by Takashi Iwai 31 Jul '18

31 Jul '18

On Tue, 31 Jul 2018 15:51:06 +0200, Hans Hu(SH-RD) wrote: > > I did the test and Oops report when install driver( both snoop and non-snoop mode) I don't understand. Did my one-liner change the behavior or not? thanks, Takashi > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 31 July 2018 19:59 > 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> > 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> > 主题: Re: [alsa-devel] A bug about cache inconsistency report > > On Tue, 31 Jul 2018 12:52:04 +0200, > Hans Hu(SH-RD) wrote: > > > > Hi All, > > > > Try send an email. > > > > Appearance: > > when use non-snoop + ioctl path combination, an issue occurs: > > usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it performance > > intermittently noise; > > > > Root case: > > Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. > > [cid:image002.png@01D428FF.92621520] > > However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. > > > > Related bug: > > SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a > > OK, so did you confirm that dropping the SG buffer support "fixes" the noise problem on your system? The easiest test would be like: > > --- a/sound/pci/hda/hda_controller.c > +++ b/sound/pci/hda/hda_controller.c > @@ -770,7 +770,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, > size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; > if (size > MAX_PREALLOC_SIZE) > size = MAX_PREALLOC_SIZE; > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > chip->card->dev, > size, MAX_PREALLOC_SIZE); > return 0; > > > thanks, > > Takashi > > > 保密声明： > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > CONFIDENTIAL NOTE: > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

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[alsa-devel] [PATCH 0/4] usb-audio: Add UAC3 Power Domains
by Jorge Sanjuan 31 Jul '18

31 Jul '18

This patchset add support for UAC3 Power Domains. This feature of the USB audio class 3 allows the host to notify the device what it is making use of so power comsumption can be optimized. This proposal implements this feature for Power Domains that include an Input/Output Terminal associated to an audio Streaming interface. This is the main usage of this feature according to the spec. For that reason, the logic for the Power Domain state change has been implemented within the ALSA PCMs logic and the suspend/resume callbacks of the usb_driver. The behaviour would be as follows: * Power Domain State D0: A Power Domain will reach this state only when the audio substream associated to that domain is being used (i,e. Audio playback/capture is happening). * Power Domain State D1: This is the Idle state where the driver is going to always want to be in order to reduce power consumption. * Power Domain State D2: This state is only set when the usb driver asumes the device is not going to be used anymore and hence, it wont care about getting any interrupts from the device. This will only happen when power level is set to "auto" in sysfs so the usb driver gets suspended when the interfaces are not in use. NOTE: The way this has been implemented will always try to put the Power Domain in state D1 if the Power Domain exists so there is not a way a user could disable this feature. It may be worth getting a control exposed to userland that enables/disables this feature (?). Power Domains affecting other units independently are required to be bypassed via a Selector Unit first before the host can change the power state. This sceneario is not covered in this patchset. based on next-20180719 Jorge Sanjuan (4): ALSA: usb-audio: Initial Power Domain support ALSA: usb-audio: AudioStreaming Power Domain parsing ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks ALSA: usb-audio: Add UAC3 Power Domains to suspend/resume include/linux/usb/audio-v3.h | 4 ++ sound/usb/Makefile | 1 + sound/usb/card.c | 9 ++++ sound/usb/card.h | 2 + sound/usb/pcm.c | 64 +++++++++++++++++++++-- sound/usb/pcm.h | 2 + sound/usb/power.c | 117 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/power.h | 19 +++++++ sound/usb/stream.c | 70 +++++++++++++++++++++++--- 9 files changed, 277 insertions(+), 11 deletions(-) create mode 100644 sound/usb/power.c -- 2.11.0

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[alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 31 Jul '18

31 Jul '18

Hi All, Try send an email. Appearance: when use non-snoop + ioctl path combination, an issue occurs: usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it performance intermittently noise; Root case: Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. [cid:image002.png@01D428FF.92621520] However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. Related bug: SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a Best Regards! Hans Hu ????? ????????????????????????????????????????????????????? CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

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[alsa-devel] More snd_pcm_ioplug_avail_update() questions
by Rob Duncan 31 Jul '18

31 Jul '18

Is it required that all available data has been committed between calls to snd_pcm_ioplug_avail_update() on an IO plugin capture stream? If this is not a requirement then there can still be uncommitted data in the mmap after the current appl_ptr. This data will get overwritten by the transfer call. If it is a requirement then snd_pcm_rate_avail_update() seems to be doing the wrong thing, because it only commits in units of the period size, which means that there could be a partial period left in the mmap. Can someone clarify what the contract is for these calls? Thanks, Rob.

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[alsa-devel] [asoc:for-4.19 199/244] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 31 Jul '18

31 Jul '18

Hi Jerome, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: d8504acca759ae672c6d34d49a33d54ace094cbb commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/244] ASoC: meson: add tdm output driver config: openrisc-allyesconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] how could unsubscribe ?
by fulgor fulgor 31 Jul '18

31 Jul '18

how could unsubscribe ?

2 1

[alsa-devel] [PATCH 0/4] ASoC: Tegra30 TDM support
by Jorge Sanjuan 30 Jul '18

30 Jul '18

This patchset adds support for TDM audio on Tegra30 hardware. It adds the DAI's `set_tdm_slot` callback and enables a tegra pcm to have up to 8 channels. It also includes support for other audio formats supported by the Tegra30 HW and fixes a broken macro needed for setting the TDM on the registers. Based on Linux 4.18-rc3 tag. Edward Cragg (4): ASoC: tegra: i2s: Fix typo/broken macro ASoC: tegra: Add a TDM configuration callback ASoC: tegra: Allow 32-bit and 24-bit samples ASoC: tegra: i2s: Add support for more than 2 channels sound/soc/tegra/tegra30_i2s.c | 72 ++++++++++++++++++++++++++++++++++++------- sound/soc/tegra/tegra30_i2s.h | 2 +- 2 files changed, 62 insertions(+), 12 deletions(-) -- 2.11.0

6 19

[alsa-devel] [PATCH 1/2] ALSA: seq: virmidi: Offload the output event processing
by Takashi Iwai 30 Jul '18

30 Jul '18

The virmidi sequencer stuff tries to translate the rawmidi bytes to sequencer events and deliver the packets at trigger callback. The amount of the whole process of these translations and deliveries depends on the incoming rawmidi bytes, and we have no limit for that; this was the cause of a CPU soft lockup that had been reported and fixed recently. Although we've fixed the soft lockup by putting the temporary unlock and cond_resched(), it's rather a quick band aid. In this patch, meanwhile, the event parsing and delivery process is offloaded to a dedicated work, and the trigger callback just kicks it off. It has three merits, at least: - The processing is always done in a sleepable context, which can assure the event delivery with non-atomic flag without hackish is_atomic() usage. - Other relevant codes can be simplified, reducing the lines - It makes me happier Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/seq_virmidi.h | 1 + sound/core/seq/seq_virmidi.c | 100 ++++++++++++++++------------------- 2 files changed, 47 insertions(+), 54 deletions(-) diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index 695257ae64ac..d488dcfa3a4e 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -41,6 +41,7 @@ struct snd_virmidi { struct snd_seq_event event; struct snd_virmidi_dev *rdev; struct snd_rawmidi_substream *substream; + struct work_struct output_work; }; #define SNDRV_VIRMIDI_SUBSCRIBE (1<<0) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 8ebbca554e99..67ea5d62cebc 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -154,70 +154,58 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i } } -/* - * trigger rawmidi stream for output +/* process rawmidi bytes and send events; + * we need no lock here for vmidi->event since it's handled only in this work */ -static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +static void snd_vmidi_output_work(struct work_struct *work) { - struct snd_virmidi *vmidi = substream->runtime->private_data; - int count, res; - unsigned char buf[32], *pbuf; - unsigned long flags; - bool check_resched = !in_atomic(); + struct snd_virmidi *vmidi; + struct snd_rawmidi_substream *substream; + unsigned char input; + int ret; + + vmidi = container_of(work, struct snd_virmidi, output_work); + substream = vmidi->substream; + + /* discard the outputs in dispatch mode unless subscribed */ + if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && + !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { + while (!snd_rawmidi_transmit_empty(substream)) + snd_rawmidi_transmit_ack(substream, 1); + return; + } - if (up) { - vmidi->trigger = 1; - if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && - !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - while (snd_rawmidi_transmit(substream, buf, - sizeof(buf)) > 0) { - /* ignored */ - } - return; - } - spin_lock_irqsave(&substream->runtime->lock, flags); + while (vmidi->trigger) { + if (snd_rawmidi_transmit(substream, &input, 1) != 1) + break; + if (snd_midi_event_encode_byte(vmidi->parser, input, + &vmidi->event) <= 0) + continue; if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; + ret = snd_seq_kernel_client_dispatch(vmidi->client, + &vmidi->event, + false, 0); vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - while (1) { - count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf)); - if (count <= 0) + if (ret < 0) break; - pbuf = buf; - while (count > 0) { - res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event); - if (res < 0) { - snd_midi_event_reset_encode(vmidi->parser); - continue; - } - __snd_rawmidi_transmit_ack(substream, res); - pbuf += res; - count -= res; - if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; - vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - } - if (!check_resched) - continue; - /* do temporary unlock & cond_resched() for avoiding - * CPU soft lockup, which may happen via a write from - * a huge rawmidi buffer - */ - spin_unlock_irqrestore(&substream->runtime->lock, flags); - cond_resched(); - spin_lock_irqsave(&substream->runtime->lock, flags); } - out: - spin_unlock_irqrestore(&substream->runtime->lock, flags); - } else { - vmidi->trigger = 0; + /* rawmidi input might be huge, allow to have a break */ + cond_resched(); } } +/* + * trigger rawmidi stream for output + */ +static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct snd_virmidi *vmidi = substream->runtime->private_data; + + vmidi->trigger = !!up; + if (up) + queue_work(system_highpri_wq, &vmidi->output_work); +} + /* * open rawmidi handle for input */ @@ -270,6 +258,7 @@ static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream) vmidi->port = rdev->port; snd_virmidi_init_event(vmidi, &vmidi->event); vmidi->rdev = rdev; + INIT_WORK(&vmidi->output_work, snd_vmidi_output_work); runtime->private_data = vmidi; return 0; } @@ -299,6 +288,9 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream) static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_virmidi *vmidi = substream->runtime->private_data; + + vmidi->trigger = 0; /* to be sure */ + cancel_work_sync(&vmidi->output_work); snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; kfree(vmidi); -- 2.18.0

1 1

[alsa-devel] [PATCH v2 0/6] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 30 Jul '18

30 Jul '18

From: Liang Chen <cl(a)rock-chips.com> Changes in V2: 1. change the subject to format "dt-bindings: module: ..." for the documents. 2. use new property <num-interpolated-steps> for backlight. 3. remove pinctrl of rk809, because the driver is not ready. Liang Chen (6): dt-bindings: usb: dwc2: add description for px30 dt-bindings: mmc: rockchip-dw-mshc: add description for px30 dt-bindings: sound: rockchip-i2s: add description for px30 dt-bindings: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/mmc/rockchip-dw-mshc.txt | 1 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 473 +++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2043 ++++++++++++++++++++ 8 files changed, 2526 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8988: fix typo in rate constraints
by Ladislav Michl 30 Jul '18

30 Jul '18

Remove duplicated entry and add missing zero in rate constraints. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- sound/soc/codecs/wm8988.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; -- 2.18.0

2 1

[alsa-devel] [PATCH] ALSA: memalloc: Fix missing PAGE_SIZE definition
by Takashi Iwai 30 Jul '18

30 Jul '18

The recent fix moved the inline snd_sgbuf_aligned_pages() outside the ifdef, and this triggered a build error on some architectures due to the undefined PAGE_SIZE, as spotted by 0day bot. Fix it by adding the missing header inclusion. Fixes: 4cae99d9b530 ("ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally") Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c669900e6cbe..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; /* -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: ak4642: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 6050559..3532370 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori(a)renesas.com> +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: ak4554: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4554.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee1340..2fa83a1 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/module.h> #include <sound/soc.h> @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: da7210: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111..e172913 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen <Dajun.chen(a)diasemi.com> - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen <Dajun.chen(a)diasemi.com> +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include <linux/delay.h> #include <linux/i2c.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: ak4613: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9..c1181a2 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include <linux/clk.h> #include <linux/delay.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: sh: Kconfig: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> By default all files without license information are under the default license of the kernel, which is GPL version 2. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800..dc20f0f 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: tegra_alc5632: fix device_node refcounting" to the asoc tree
by Mark Brown 30 Jul '18

30 Jul '18

The patch ASoC: tegra_alc5632: fix device_node refcounting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4321723648b0abb456f7a9af51bb09a4ec60799d Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Date: Sat, 28 Jul 2018 00:06:59 +0300 Subject: [PATCH] ASoC: tegra_alc5632: fix device_node refcounting tegra_alc5632_probe() increments reference count of device nodes with of_parse_phandle(), but there is no code decrementing them in the driver. The patch adds of_node_put() to tegra_alc5632_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_alc5632.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } -- 2.18.0

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[alsa-devel] [PATCH] ASoC: sirf: Fix potential NULL pointer dereference
by Gustavo A. R. Silva 30 Jul '18

30 Jul '18

There is a potential execution path in which function platform_get_resource() returns NULL. If this happens, we will end up having a NULL pointer dereference. Fix this by replacing devm_ioremap with devm_ioremap_resource, which has the NULL check and the memory region request. This code was detected with the help of Coccinelle. Cc: stable(a)vger.kernel.org Fixes: 2bd8d1d5cf89 ("ASoC: sirf: Add audio usp interface driver") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sirf/sirf-usp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf..d70fcd4 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: uniphier: add support for multichannel output
by Katsuhiro Suzuki 30 Jul '18

30 Jul '18

This patch adds multichannel PCM output support for LD11/LD20. Currently driver tested and supported only 2ch, 6ch, and 8ch. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 78 ++++++++++++++++++++++++++++++++--- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-reg.h | 1 + sound/soc/uniphier/aio.h | 3 ++ 4 files changed, 78 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..8b09bbb0f8d0 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -264,6 +264,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } } +/** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + /** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..23a5c3c68658 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * -- 2.18.0

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[alsa-devel] [PATCH] ASoC: uniphier: change functions to static
by Katsuhiro Suzuki 30 Jul '18

30 Jul '18

This patch changes some functions that are not used by other objects to static. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 6 +++--- sound/soc/uniphier/aio.h | 3 --- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 8b09bbb0f8d0..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -327,7 +327,7 @@ static int aio_port_set_ch(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -446,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -511,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 23a5c3c68658..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -325,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); -- 2.18.0

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