June 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: da7219: Fix build error without CONFIG_I2C
by YueHaibing 04 Jun '19

04 Jun '19

Fix gcc build error while CONFIG_I2C is not set sound/soc/codecs/da7219.c:2640:1: warning: data definition has no type or storage class module_i2c_driver(da7219_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/da7219.c:2640:1: error: type defaults to int in declaration of module_i2c_driver [-Werror=implicit-int] sound/soc/codecs/da7219.c:2640:1: warning: parameter names (without types) in function declaration sound/soc/codecs/da7219.c:2629:26: warning: da7219_i2c_driver defined but not used [-Wunused-variable] Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 6d817c0e9fd7 ("ASoC: codecs: Add da7219 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 933ab51..111e44b 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -133,7 +133,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A config SND_SOC_MT8183_DA7219_MAX98357A tristate "ASoC Audio driver for MT8183 with DA7219 MAX98357A codec" - depends on SND_SOC_MT8183 + depends on SND_SOC_MT8183 && I2C select SND_SOC_MT6358 select SND_SOC_MAX98357A select SND_SOC_DA7219 -- 2.7.4

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[alsa-devel] [PATCH v2 20/22] dt: bindings: fix some broken links from txt->yaml conversion
by Mauro Carvalho Chehab 04 Jun '19

04 Jun '19

Some new files got converted to yaml, but references weren't updated accordingly. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- Documentation/devicetree/bindings/media/st,stm32-dcmi.txt | 2 +- Documentation/devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- MAINTAINERS | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt b/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt index 249790a93017..3122ded82eb4 100644 --- a/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt +++ b/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt @@ -11,7 +11,7 @@ Required properties: - clock-names: must contain "mclk", which is the DCMI peripherial clock - pinctrl: the pincontrol settings to configure muxing properly for pins that connect to DCMI device. - See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt. + See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml. - dmas: phandle to DMA controller node, see Documentation/devicetree/bindings/dma/stm32-dma.txt - dma-names: must contain "tx", which is the transmit channel from DCMI to DMA diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt index 58c341300552..cbf24bcd1b8d 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -18,7 +18,7 @@ Required properties: See Documentation/devicetree/bindings/dma/stm32-dma.txt. - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml Optional properties: - resets: Reference to a reset controller asserting the reset controller diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index 3f4467ff0aa2..944743dd9212 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -41,7 +41,7 @@ SAI subnodes required properties: "tx": if sai sub-block is configured as playback DAI "rx": if sai sub-block is configured as capture DAI - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml SAI subnodes Optional properties: - st,sync: specify synchronization mode. diff --git a/MAINTAINERS b/MAINTAINERS index 0dc7c3c5ddb0..2ab2337a029c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1298,7 +1298,7 @@ ARM PRIMECELL SSP PL022 SPI DRIVER M: Linus Walleij <linus.walleij(a)linaro.org> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/spi/spi_pl022.txt +F: Documentation/devicetree/bindings/spi/spi-pl022.yaml F: drivers/spi/spi-pl022.c ARM PRIMECELL UART PL010 AND PL011 DRIVERS -- 2.21.0

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[alsa-devel] [PATCH v3] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Change-Id: I304882e0b67974f8fe4c2a47c61a41a04635b2df Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- Fix compile error. codec->dai => dai runtime => substream->runtime sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..ada8c25e643d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

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[alsa-devel] [PATCH v2] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..a6c2cb89767c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

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[alsa-devel] [PATCH] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..4cbfb0b95404 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if RJ is 1. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

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[alsa-devel] [PATCH 0/5] Assorted fixes discovered with gcc 4.1
by Geert Uytterhoeven 04 Jun '19

04 Jun '19

Hi all, Ever since commit cafa0010cd51fb71 ("Raise the minimum required gcc version to 4.6"), I felt bored when looking at my test build logs, as I was no longer discovering many real issues. Hence I started wondering if the modern gcc versions are really catching these classes of bugs caught before with gcc 4.1, or if they just go undetected. I reverted some changes and applied some fixes, which allowed me to compile most of the kernel with gcc 4.1 again. I built an m68k/allmodconfig kernel, looked at all new warnings, and fixed the ones that are not false positives. The result is a patch series of 5 patches, of which one or two fix real bugs. Thanks for your comments, and for applying where appropriate! Geert Uytterhoeven (5): lightnvm: Fix uninitialized pointer in nvm_remove_tgt() rxrpc: Fix uninitialized error code in rxrpc_send_data_packet() net: sched: pie: Use ULL suffix for 64-bit constant ALSA: fireface: Use ULL suffixes for 64-bit constants [RFC] devlink: Fix uninitialized error code in devlink_fmsg_prepare_skb() drivers/lightnvm/core.c | 2 +- net/core/devlink.c | 2 +- net/rxrpc/output.c | 4 +++- net/sched/sch_pie.c | 2 +- sound/firewire/fireface/ff-protocol-latter.c | 10 +++++----- 5 files changed, 11 insertions(+), 9 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Re: [alsa-devel] atmel SSC DSP A mode
by Codrin.Ciubotariu＠microchip.com 04 Jun '19

04 Jun '19

On 03.06.2019 13:19, Willem Romkes wrote: > > * > * > > Hi, > > We are using linux-at91 and lately I have been working on adding support > for a custom soundcard, a dual tlv320aic3x codec on SAMA5D3 platform. > This soundcard uses DSP A mode. > > We are facing audio issues: When we playback a WAV file with a sine > wave, the codec outputs far from a clean sine wave. The problem I think > is that the CKI settings for the TMCR register isn't set correctly. > see commit 20cf2603b122bf71fb54def1de6a2ad73d5ddb0b > Author: Bo Shen <voice.shen(a)atmel.com> > Date: Fri Jan 30 17:38:42 2015 +0800 > > ASoC: atmel_ssc_dai: fix the setting for DSP mode > > When SCC work in DSP A mode, the data outputs/inputs are shift out on > falling edge, the frame sync are sample on the rising edge. > > Reported-by: Songjun Wu <songjun.wu(a)atmel.com> > Signed-off-by: Bo Shen <voice.shen(a)atmel.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/atmel/atmel_ssc_dai.c > b/sound/soc/atmel/atmel_ssc_dai.c > index e691aab..35e44e4 100644 > --- a/sound/soc/atmel/atmel_ssc_dai.c > +++ b/sound/soc/atmel/atmel_ssc_dai.c > @@ -502,7 +502,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) > | SSC_BF(RCMR_STTDLY, 1) > | SSC_BF(RCMR_START, SSC_START_RISING_RF) > - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(RCMR_CKO, SSC_CKO_NONE) > | SSC_BF(RCMR_CKS, SSC_CKS_DIV); > > @@ -517,7 +517,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) > | SSC_BF(TCMR_STTDLY, 1) > | SSC_BF(TCMR_START, SSC_START_RISING_RF) > - | SSC_BF(TCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) > | SSC_BF(TCMR_CKS, SSC_CKS_DIV); > > @@ -546,7 +546,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > rcmr = SSC_BF(RCMR_PERIOD, 0) > | SSC_BF(RCMR_STTDLY, START_DELAY) > | SSC_BF(RCMR_START, SSC_START_RISING_RF) > - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(RCMR_CKO, SSC_CKO_NONE) > | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? > SSC_CKS_PIN : SSC_CKS_CLOCK); > > For both the TCMR and RCMR the CKI is set to SSC_CKI_FALLING, but I > think for the TCMR the CKI should be set to SSC_CKI_RISING. See datasheet: > > TCMR CKI: "1: The data outputs (Data and Frame Sync signals) are shifted > out on Transmit Clock rising edge. The Frame sync signal > input is sampled on Transmit clock falling edge." > > RCMR CKI: "0: The data inputs (Data and Frame Sync signals) are sampled > on Receive Clock falling edge. The Frame Sync signal out- > put is shifted out on Receive Clock rising edge." > > Setting the CKI in TCMR to SSC_CKI_RISING to fixes our issues. Hi Willem, It this DS of the codec you are using? http://www.ti.com/lit/ds/symlink/tlv320aic33.pdf Are you using the DSP mode described in figure 25 DSP Serial Bus Mode Operation? If yes, then you should use the DSP_B mode... Thanks and best regards, Codrin

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[alsa-devel] [PATCH] ASoC: core: move DAI pre-links initiation to snd_soc_instantiate_card
by Tzung-Bi Shih 04 Jun '19

04 Jun '19

Kernel crashes when an ASoC component rebinding. As an example, by using the following commands: - echo -n max98357a > /sys/bus/platform/drivers/max98357a/unbind - echo -n max98357a > /sys/bus/platform/drivers/max98357a/bind Got the error message: "Unable to handle kernel NULL pointer dereference at virtual address". The call trace: snd_soc_is_matching_component+0x30/0x6c soc_bind_dai_link+0x16c/0x240 snd_soc_bind_card+0x1e4/0xb10 snd_soc_add_component+0x270/0x300 snd_soc_register_component+0x54/0x6c devm_snd_soc_register_component+0x68/0xac max98357a_platform_probe+0x7c/0x94 The dai_link->platforms has been reset to NULL by soc_cleanup_platform() in soc_cleanup_card_resources() when un-registering component. However, it has no chance to re-allocate the dai_link->platforms when registering the component again. Move the DAI pre-links initiation from snd_soc_register_card() to snd_soc_instantiate_card() to make sure all DAI pre-links get initiated when component rebinding. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-core.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d3520fca613..82ff384753c7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2072,6 +2072,15 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + for_each_card_prelinks(card, i, dai_link) { + ret = soc_init_dai_link(card, dai_link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init link %s: %d\n", + dai_link->name, ret); + goto probe_end; + } + } + card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; @@ -2241,7 +2250,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_sync(&card->dapm); probe_end: - if (ret < 0) + if (ret < 0 && ret != -EPROBE_DEFER) soc_cleanup_card_resources(card); mutex_unlock(&card->mutex); @@ -2794,26 +2803,9 @@ static int snd_soc_bind_card(struct snd_soc_card *card) */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; - struct snd_soc_dai_link *link; - if (!card->name || !card->dev) return -EINVAL; - mutex_lock(&client_mutex); - for_each_card_prelinks(card, i, link) { - - ret = soc_init_dai_link(card, link); - if (ret) { - soc_cleanup_platform(card); - dev_err(card->dev, "ASoC: failed to init link %s\n", - link->name); - mutex_unlock(&client_mutex); - return ret; - } - } - mutex_unlock(&client_mutex); - dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 2.22.0.rc1.257.g3120a18244-goog

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[alsa-devel] [PATCH 0/5] ASoC: SOF: suspend/resume debug tools
by Pierre-Louis Bossart 03 Jun '19

03 Jun '19

No new functionality per se, just utilities for developers. The vast majority of current integration issues are related to HDAudio + trace support combined with runtime suspend/resume. Add a couple of changes to make it easier for developers to see the firmware logs across suspend/resume transitions, disable the trace and/or disable runtime_pm to help narrow down the problems. Kai Vehmanen (2): ASoC: SOF: trace: remove code duplication in sof_wait_trace_avail() ASoC: SOF: force end-of-file for debugfs trace at suspend Pierre-Louis Bossart (3): ASoC: SOF: trace: move to opt-in with Kconfig and module parameter ASoC: SOF: acpi: add module param to disable pm_runtime ASoC: SOF: pci: add module param to disable pm_runtime sound/soc/sof/Kconfig | 8 +++++ sound/soc/sof/core.c | 26 ++++++++++---- sound/soc/sof/sof-acpi-dev.c | 12 ++++++- sound/soc/sof/sof-pci-dev.c | 12 ++++++- sound/soc/sof/sof-priv.h | 3 ++ sound/soc/sof/trace.c | 67 ++++++++++++++++++++++++++++++------ 6 files changed, 110 insertions(+), 18 deletions(-) base-commit: 188d45fe779eeb8e3521b59fcb12cc48a6f2c203 -- 2.20.1

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[alsa-devel] [PATCH 0/9] ASoC: SOF: new functionality with MINOR ABI changes
by Pierre-Louis Bossart 03 Jun '19

03 Jun '19

The first patches are to align with firmware, we unfortunately had a couple of misses before the new ABI change process was introduced [1]. The new functionality added after the initial cleanup is: 1. sending the kernel time to firmware so that the trace logs use the same base across suspend-resume cycles. 2. introducing an IPC flood mechanism which already helped harden the IPC sequences, be it with QEMU or on real hardware. 3. support for mux/demux component 4. initial SoundWire support This patchset is independent from the IPC changes submitted on alsa-devel, but will only apply if previous trace related changes are applied. ([PATCH 3/5] ASoC: SOF: trace: move to opt-in with Kconfig and module parameter) [1] https://github.com/orgs/thesofproject/projects/2 Bard liao (1): ASoC: SOF: send time stamp to FW for alignment Jaska Uimonen (1): ASoC: SOF: topology: add support for mux/demux component Kai Vehmanen (1): ASoC: SOF: fix DSP oops definitions in FW ABI Pan Xiuli (1): ASoC: SOF: soundwire: add initial soundwire support Pierre-Louis Bossart (2): ASoC: SOF: uapi: mirror firmware changes ASoC: SOF: bump to ABI 3.6 Ranjani Sridharan (3): ASoC: SOF: add mode parameter for snd_sof_debugfs_buf_item ASoC: SOF: ipc: Introduce SOF_IPC_GLB_TEST_MSG IPC command ASoC: SOF: debug: add new debugfs entries for IPC flood test include/sound/sof/dai.h | 1 + include/sound/sof/header.h | 30 ++++- include/sound/sof/topology.h | 3 + include/sound/sof/trace.h | 10 ++ include/sound/sof/xtensa.h | 9 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/Kconfig | 8 ++ sound/soc/sof/debug.c | 247 ++++++++++++++++++++++++++++++++++- sound/soc/sof/intel/bdw.c | 17 ++- sound/soc/sof/intel/byt.c | 15 ++- sound/soc/sof/intel/hda.c | 16 ++- sound/soc/sof/ipc.c | 9 ++ sound/soc/sof/loader.c | 4 +- sound/soc/sof/sof-priv.h | 8 +- sound/soc/sof/topology.c | 2 + sound/soc/sof/trace.c | 16 ++- sound/soc/sof/xtensa/core.c | 2 +- 17 files changed, 366 insertions(+), 33 deletions(-) -- 2.20.1

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