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

[alsa-devel] [PATCH] ASoC:hdac_hda: use correct format to setup hda codec
by Rander Wang 25 Feb '19

25 Feb '19

The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.14.1

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[alsa-devel] Merge window for 5.1
by Takashi Iwai 25 Feb '19

25 Feb '19

Hi, I'll be traveling in the whole next week (from March 3 to 11), and likely offline during that. This schedule will conflict with the first week of the planned 5.1 merge window, so I'd like to close the merge a bit earlier than usual at this time. That is, if anyone has some urgent patch to be merged for 5.1, please submit ASAP for review. Mark, could you prepare the ASoC merge materials until Thursday? Then I'll send an early pull request to Linus on the upcoming weekend. Small fixes can be covered in the second week of the merge window, we should push out the major changes beforehand. thanks, Takashi

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[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add Maxim98373 support
by Jenny TC 25 Feb '19

25 Feb '19

Reuse kbl_da7219_max98927 machine driver to support max98373 v1: First patchset V2: Updated commit message and Acked-by Jenny TC (1): ASoC: Intel: Boards: Add Maxim98373 support sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 203 +++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 19 ++ 3 files changed, 200 insertions(+), 23 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda: Extend i915 component bind timeout
by Takashi Iwai 25 Feb '19

25 Feb '19

I set 10 seconds for the timeout of the i915 audio component binding with a hope that recent machines are fast enough to handle all probe tasks in that period, but I was too optimistic. The binding may take longer than that, and this caused a problem on the machine with both audio and graphics driver modules loaded in parallel, as Paul Menzel experienced. This problem haven't hit so often just because the KMS driver is loaded in initrd on most machines. As a simple workaround, extend the timeout to 60 seconds. Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Reported-by: Paul Menzel <pmenzel+alsa-devel(a)molgen.mpg.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 617ff1aa818f..27eb0270a711 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -144,9 +144,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) return -ENODEV; if (!acomp->ops) { request_module("i915"); - /* 10s timeout */ + /* 60s timeout */ wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(10 * 1000)); + msecs_to_jiffies(60 * 1000)); } if (!acomp->ops) { dev_info(bus->dev, "couldn't bind with audio component\n"); -- 2.16.4

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[alsa-devel] A bug in the alsa project
by Радослав Ненчовски 25 Feb '19

25 Feb '19

*Hi. I think I found a bug in the ALSA project which doesn't seem to be kernel related, since it occurs with all kernels I've tried so far. It's about the area where one can select the output - Stereo, 2.1, 4.0 and so on. I have a 2.1 audio system - 2 speakers and a subwoofer (Microlab M-300U). If I choose 5.1 Analog Output, the subwoofer works great and the audio has great bass for music. However, if I choose anything else that is not 5.1 (that includes the output for my system - 2.1), the subwoofer stops working and the output sounds like I'm using cheap speakers for $1. This is part 1 of the bug. Part 2 is that when I'm watching movies using the 5.1 Output, some of the movie sounds are missing because ALSA transfers them to the non-existing speakers. It's also interesting that with 5.1 Analog Output listening to music (Audacious) the sound is perfect - no missing parts and stuff, the problem happens only when I'm watching movies. I tried different media players but it's the same with all of them.* *If there's any additional information I could give you, such as hardware info, please let me know.*

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Re: [alsa-devel] [PATCH -next] platform/chrome: Fix Kconfig dependencies for wilco_ec
by Enric Balletbo i Serra 25 Feb '19

25 Feb '19

Hi Randy, Thanks for the report On 22/2/19 17:50, Randy Dunlap wrote: > On 2/20/19 11:57 PM, Enric Balletbo i Serra wrote: >> Hi, >> >> On 21/2/19 0:09, Randy Dunlap wrote: >>> On 2/20/19 2:11 PM, Nick Crews wrote: >>>> In the initial version of the Wilco EC Driver, the >>>> dependency order was wrong. It before was possible to >>>> select CONFIG_WILCO_EC and CONFIG_CROS_EC_LPC without >>>> having CONFIG_CROS_EC_LPC_MEC. This was wrong, since >>>> WILCO_EC depends upon CONFIG CROS_EC_LPC_MEC, not the >>>> other way around. >>>> >>>> Fixes: 1733c32834e5d1 ("platform/chrome: Add new driver for Wilco EC") >>>> Signed-off-by: Nick Crews <ncrews(a)chromium.org> >>> >>> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> >>> Acked-by: Randy Dunlap <rdunlap(a)infradead.org> # build-tested >>> >> >> As this is [-next] material I squashed that commit and queued for 5.1 > > Hi, > > on linux-next of 20190222, I now see this build error + Kconfig warnings: > > (this is on i386; randconfig file is attached) > > WARNING: unmet direct dependencies detected for SND_SOC_CROS_EC_CODEC > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_CROS_EC [=n] > Selected by [y]: > - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] This error seems to be introduced by following commit. commit 74c6ecf4194ebed285b29964a950e0cd7414fe19 Author: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Mon Feb 18 12:18:19 2019 +0800 ASoC: qcom: Kconfig: select dmic for sdm845 sdm845 uses dmic on EC so it should select CROS_EC_CODEC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> The problem is that this commit selects CROS_EC_CODEC but the CROS_EC_CODEC depends on MFD_CROS_EC which is not selected. A question for Cheng-Yi, I suppose that the answer is yes but worth ask because I don't know about this device. Is the sdm845 a chromebook with and embedded controller right? Thanks, Enric > > ld: sound/soc/codecs/cros_ec_codec.o: in function `ec_command_no_resp': > cros_ec_codec.c:(.text+0x50): undefined reference to `cros_ec_cmd_xfer_status' > ld: sound/soc/codecs/cros_ec_codec.o: in function `mic_gain_get': > cros_ec_codec.c:(.text+0x5a8): undefined reference to `cros_ec_cmd_xfer_status' > > >

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[alsa-devel] [PATCH v4 00/14] ASoC: Sound Open Firmware (SOF) core
by Pierre-Louis Bossart 25 Feb '19

25 Feb '19

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core and utilities. Support for Intel devices is provided as a follow-up series. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since v3: Addressed dozens ofcomments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Hardened memory allocation, fixed module load/unload oops or errors (now at hundreds of cycles on ApolloLake devices) Fixed suspend/resume and removed use of suspend_late, now using snd_soc_pm_ops and standard flow Fixed topology free (multiple patches already contributed for ASoC-core) Fixed debugfs/pm_runtime interaction Removed error checks on debugfs, changed to EXPORT_SYMBOL_GPL Added filenames in debug messages, ues hex_dump_to_buffer Removed "sof-audio" platform device and added better error checks on probe Reworked data structures to remove mix of const/variable fields and duplication of pointers for platform-specific changes Removed redundant ops tables Fixed trace/logger issues Fixed issues with loader (alignment and block size errors) Fixed release_firmware handling (was all over the place) Simplified PCI handling Added comments on non-atomic triggers Removed enum controls (not supported in topology) Simplified include file dependencies Fix cppcheck warnings Fixed Kconfigs to deal with Kbuild warnings on exotic architectures Precisions: The code in this patchset is directly squashed from the SOF development branch [1], which tracks Mark Brown's for-next branch on a weekly basis and the configurations used for testing are based on the defconfigs at [2]. The patches are also used backported for Chromebook devices. Full disclosure on known limitation and issues (full list at [3]) a) the interaction with ASoC is not always perfect, there are a couple of points where a better solution is desired (use of private data, support for link DMA, etc). The known points are explicitly documented in the code and will be updated. To be clearer, it's our belief that SOF does not cripple ASoC in any way, and that merging this SOF core does not harm others. It's just complicated to get things right, especially on a couple of cases where the topology and DPCM frameworks generate complex flows that very few people understand. b) The get/put methods for controls generate an IPC and possibly a wake-up. This is not optimal and is being fixed. c) runtime_pm is being hardened and the use of SMART_SUSPEND was suggested. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. This patchset also includes contributions from Daniel Baluta for loading code on non-Intel platforms. Pierre [1] https://github.com/thesofproject/linux [2] https://github.com/thesofproject/kconfig [3] https://github.com/thesofproject/linux/issues Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support Pierre-Louis Bossart (2): ASoC: SOF: Add xtensa support ASoC: SOF: Add utils include/sound/soc.h | 3 + include/sound/sof.h | 100 + include/sound/sof/control.h | 125 ++ include/sound/sof/dai-intel.h | 178 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 ++ include/sound/sof/info.h | 118 ++ include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 149 ++ include/sound/sof/topology.h | 256 +++ include/sound/sof/trace.h | 66 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 62 + include/uapi/sound/sof/eq.h | 164 ++ include/uapi/sound/sof/fw.h | 78 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 98 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 93 + sound/soc/sof/control.c | 361 ++++ sound/soc/sof/core.c | 473 +++++ sound/soc/sof/debug.c | 219 +++ sound/soc/sof/ipc.c | 842 +++++++++ sound/soc/sof/loader.c | 363 ++++ sound/soc/sof/nocodec.c | 109 ++ sound/soc/sof/ops.c | 204 +++ sound/soc/sof/ops.h | 414 +++++ sound/soc/sof/pcm.c | 710 ++++++++ sound/soc/sof/pm.c | 381 ++++ sound/soc/sof/sof-priv.h | 599 ++++++ sound/soc/sof/topology.c | 2808 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 298 +++ sound/soc/sof/utils.c | 109 ++ sound/soc/sof/xtensa/Kconfig | 2 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 138 ++ 37 files changed, 10086 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c -- 2.17.1

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[alsa-devel] [PATCH] arm64: dts: renesas: r8a7796: remove unneeded sound #address/size-cells
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> commit 78bc93b3ffb2 ("arm64: dts: renesas: r8a7796: Add address properties to rcar_sound port nodes") added missing #address-cells and #size-cells for sound ports. But, these are based on platform, not on SoC. This patch cleanups it. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts | 1 + arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts | 1 + arch/arm64/boot/dts/renesas/r8a7796.dtsi | 11 ----------- 3 files changed, 2 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts b/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts index b4f9567..2aefa53 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts +++ b/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts @@ -68,6 +68,7 @@ ports { /* rsnd_port0 is on salvator-common */ rsnd_port1: port@1 { + reg = <1>; rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; diff --git a/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts b/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts index 8860be6..fa4ef7c 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts +++ b/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts @@ -69,6 +69,7 @@ ports { /* rsnd_port0 is on salvator-common */ rsnd_port1: port@1 { + reg = <1>; rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index 0648d12..cc601d9 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -2162,17 +2162,6 @@ dma-names = "rx", "tx", "rxu", "txu"; }; }; - - ports { - #address-cells = <1>; - #size-cells = <0>; - port@0 { - reg = <0>; - }; - port@1 { - reg = <1>; - }; - }; }; audma0: dma-controller@ec700000 { -- 2.7.4

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[alsa-devel] [PATCH] arm64: dts: renesas: r8a77965: add SSIU support for sound
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> rsnd driver supports SSIU now, let's use it. Then, BUSIF DMA settings on rcar_sound,ssi (= rxu, txu) are no longer needed. Applies commit 8d14bfa074db ("arm64: dts: renesas: r8a7796: add SSIU support for sound") for r8a77965. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 211 ++++++++++++++++++++++++++++++ 1 file changed, 211 insertions(+) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index 4b3730f..cc77c2e 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1585,6 +1585,217 @@ }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + ssiu02: ssiu-2 { + dmas = <&audma0 0x37>, <&audma1 0x38>; + dma-names = "rx", "tx"; + }; + ssiu03: ssiu-3 { + dmas = <&audma0 0x47>, <&audma1 0x48>; + dma-names = "rx", "tx"; + }; + ssiu04: ssiu-4 { + dmas = <&audma0 0x3F>, <&audma1 0x40>; + dma-names = "rx", "tx"; + }; + ssiu05: ssiu-5 { + dmas = <&audma0 0x43>, <&audma1 0x44>; + dma-names = "rx", "tx"; + }; + ssiu06: ssiu-6 { + dmas = <&audma0 0x4F>, <&audma1 0x50>; + dma-names = "rx", "tx"; + }; + ssiu07: ssiu-7 { + dmas = <&audma0 0x53>, <&audma1 0x54>; + dma-names = "rx", "tx"; + }; + ssiu10: ssiu-8 { + dmas = <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx"; + }; + ssiu11: ssiu-9 { + dmas = <&audma0 0x4B>, <&audma1 0x4C>; + dma-names = "rx", "tx"; + }; + ssiu12: ssiu-10 { + dmas = <&audma0 0x57>, <&audma1 0x58>; + dma-names = "rx", "tx"; + }; + ssiu13: ssiu-11 { + dmas = <&audma0 0x59>, <&audma1 0x5A>; + dma-names = "rx", "tx"; + }; + ssiu14: ssiu-12 { + dmas = <&audma0 0x5F>, <&audma1 0x60>; + dma-names = "rx", "tx"; + }; + ssiu15: ssiu-13 { + dmas = <&audma0 0xC3>, <&audma1 0xC4>; + dma-names = "rx", "tx"; + }; + ssiu16: ssiu-14 { + dmas = <&audma0 0xC7>, <&audma1 0xC8>; + dma-names = "rx", "tx"; + }; + ssiu17: ssiu-15 { + dmas = <&audma0 0xCB>, <&audma1 0xCC>; + dma-names = "rx", "tx"; + }; + ssiu20: ssiu-16 { + dmas = <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx"; + }; + ssiu21: ssiu-17 { + dmas = <&audma0 0x67>, <&audma1 0x68>; + dma-names = "rx", "tx"; + }; + ssiu22: ssiu-18 { + dmas = <&audma0 0x6B>, <&audma1 0x6C>; + dma-names = "rx", "tx"; + }; + ssiu23: ssiu-19 { + dmas = <&audma0 0x6D>, <&audma1 0x6E>; + dma-names = "rx", "tx"; + }; + ssiu24: ssiu-20 { + dmas = <&audma0 0xCF>, <&audma1 0xCE>; + dma-names = "rx", "tx"; + }; + ssiu25: ssiu-21 { + dmas = <&audma0 0xEB>, <&audma1 0xEC>; + dma-names = "rx", "tx"; + }; + ssiu26: ssiu-22 { + dmas = <&audma0 0xED>, <&audma1 0xEE>; + dma-names = "rx", "tx"; + }; + ssiu27: ssiu-23 { + dmas = <&audma0 0xEF>, <&audma1 0xF0>; + dma-names = "rx", "tx"; + }; + ssiu30: ssiu-24 { + dmas = <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx"; + }; + ssiu31: ssiu-25 { + dmas = <&audma0 0x21>, <&audma1 0x22>; + dma-names = "rx", "tx"; + }; + ssiu32: ssiu-26 { + dmas = <&audma0 0x23>, <&audma1 0x24>; + dma-names = "rx", "tx"; + }; + ssiu33: ssiu-27 { + dmas = <&audma0 0x25>, <&audma1 0x26>; + dma-names = "rx", "tx"; + }; + ssiu34: ssiu-28 { + dmas = <&audma0 0x27>, <&audma1 0x28>; + dma-names = "rx", "tx"; + }; + ssiu35: ssiu-29 { + dmas = <&audma0 0x29>, <&audma1 0x2A>; + dma-names = "rx", "tx"; + }; + ssiu36: ssiu-30 { + dmas = <&audma0 0x2B>, <&audma1 0x2C>; + dma-names = "rx", "tx"; + }; + ssiu37: ssiu-31 { + dmas = <&audma0 0x2D>, <&audma1 0x2E>; + dma-names = "rx", "tx"; + }; + ssiu40: ssiu-32 { + dmas = <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx"; + }; + ssiu41: ssiu-33 { + dmas = <&audma0 0x17>, <&audma1 0x18>; + dma-names = "rx", "tx"; + }; + ssiu42: ssiu-34 { + dmas = <&audma0 0x19>, <&audma1 0x1A>; + dma-names = "rx", "tx"; + }; + ssiu43: ssiu-35 { + dmas = <&audma0 0x1B>, <&audma1 0x1C>; + dma-names = "rx", "tx"; + }; + ssiu44: ssiu-36 { + dmas = <&audma0 0x1D>, <&audma1 0x1E>; + dma-names = "rx", "tx"; + }; + ssiu45: ssiu-37 { + dmas = <&audma0 0x1F>, <&audma1 0x20>; + dma-names = "rx", "tx"; + }; + ssiu46: ssiu-38 { + dmas = <&audma0 0x31>, <&audma1 0x32>; + dma-names = "rx", "tx"; + }; + ssiu47: ssiu-39 { + dmas = <&audma0 0x33>, <&audma1 0x34>; + dma-names = "rx", "tx"; + }; + ssiu50: ssiu-40 { + dmas = <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx"; + }; + ssiu60: ssiu-41 { + dmas = <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx"; + }; + ssiu70: ssiu-42 { + dmas = <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx"; + }; + ssiu80: ssiu-43 { + dmas = <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx"; + }; + ssiu90: ssiu-44 { + dmas = <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx"; + }; + ssiu91: ssiu-45 { + dmas = <&audma0 0x7F>, <&audma1 0x80>; + dma-names = "rx", "tx"; + }; + ssiu92: ssiu-46 { + dmas = <&audma0 0x81>, <&audma1 0x82>; + dma-names = "rx", "tx"; + }; + ssiu93: ssiu-47 { + dmas = <&audma0 0x83>, <&audma1 0x84>; + dma-names = "rx", "tx"; + }; + ssiu94: ssiu-48 { + dmas = <&audma0 0xA3>, <&audma1 0xA4>; + dma-names = "rx", "tx"; + }; + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 25 Feb '19

25 Feb '19

From: Shengjiu Wang <shengjiu.wang(a)freescale.com> There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the begginning with the normal enable flow. This is hardware issue, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..23bd0ad4ac31 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +912,12 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- 1.9.1

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