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

[alsa-devel] [PATCH] ASoC: pxa-ssp: add support for an external clock in devicetree
by Daniel Mack 05 Jul '18

05 Jul '18

Allow setting a clock called 'extclk' in the device of the ssp-dai device. If specified, this clock will be set to the mclk rate from the DAI's .set_sysclk() callback. The DAI will also configure itself to use that external clock. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/mrvl,pxa-ssp.txt | 8 ++++++ sound/soc/pxa/pxa-ssp.c | 25 +++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt index efd3fb1f68d2..feef39b4a4fd 100644 --- a/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt +++ b/Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt @@ -5,6 +5,14 @@ Required properties: compatible Must be "mrvl,pxa-ssp-dai" port A phandle reference to a PXA ssp upstream device +Optional properties: + + clock-names + clocks Through "clock-names" and "clocks", external clocks + can be configured. If a clock names "extclk" exists, + it will be set to the mclk rate of the audio stream + and be used as clock provider of the DAI. + Example: /* upstream device */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..c43fe3b9c79c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -41,6 +41,7 @@ */ struct ssp_priv { struct ssp_device *ssp; + struct clk *extclk; unsigned long ssp_clk; unsigned int sysclk; unsigned int dai_fmt; @@ -205,6 +206,21 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (priv->extclk) { + int ret; + + /* + * For DT based boards, if an extclk is given, use it + * here and configure PXA_SSP_CLK_EXT. + */ + + ret = clk_set_rate(priv->extclk, freq); + if (ret < 0) + return ret; + + clk_id = PXA_SSP_CLK_EXT; + } + dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); @@ -773,6 +789,15 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai) ret = -ENODEV; goto err_priv; } + + priv->extclk = devm_clk_get(dev, "extclk"); + if (IS_ERR(priv->extclk)) { + ret = PTR_ERR(priv->extclk); + if (ret == -EPROBE_DEFER) + return ret; + + priv->extclk = NULL; + } } else { priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { -- 2.17.1

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[alsa-devel] [PATCH v3] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 04 Jul '18

04 Jul '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..82a41e4d7170 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; + /* use wait time from substream if available */ + if (substream->wait_time) + wait_time = substream->wait_time; + else + wait_time = 10 * 1000; /* 10 secs */ + if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->period_size * 2 / + (runtime->rate / 1000); wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time = msecs_to_jiffies(wait_time); } for (;;) { -- 2.17.1

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[alsa-devel] [PATCH] ASoC: nau8825: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 04 Jul '18

04 Jul '18

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Notice that such constant is used in a context that expects an expression of type u64 (64 bits, unsigned) and the following expression is currently being evaluated using 32-bit arithmetic: 256 * fs * 2 * mclk_src_scaling[i].param Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dc6ea49..b9fed99 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2016,7 +2016,7 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; -- 2.7.4

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[alsa-devel] [PATCH 0/4] ASoC: Use snd_pcm_stop_xrun() helper
by Takashi Iwai 04 Jul '18

04 Jul '18

Hi, this is a series of trivial fixes to replace some open codes with the standard snd_pcm_stop_xrun() helper. Functionally same, but it addresses the potential stream lock issues automagically, too. Takashi === Takashi Iwai (4): ASoC: davinci: Use snd_pcm_stop_xrun() helper ASoC: qcom: Use snd_pcm_stop_xrun() helper ASoC: sti: Use snd_pcm_stop_xrun() helper ASoC: stm32: Use snd_pcm_stop_xrun() helper sound/soc/davinci/davinci-mcasp.c | 16 ++++------------ sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 ++----- 5 files changed, 11 insertions(+), 22 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 0/6] ASoC: qdsp6: Add missing slimbus capture support
by Srinivas Kandagatla 04 Jul '18

04 Jul '18

This patchset adds support to slimbus tx ports plus 3 trival fixes to do with closing port when its not really open. slim tx support invoving add dais and routing, rest of the dsp configuration commands are already available in mainline. slim tx is verified with wcd9335 codec on DB820c. Thanks, srini Srinivas Kandagatla (6): ASoC: qdsp6: q6afe: Add missing slimbus capture ports ASoC: qdsp6: q6afe-dai: add support to slim tx dais ASoC: qdsp6: q6routing: add slim rx routings ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels ASoC: qdsp6: q6afe-dai: do not close port if its not opened ASoC: qdsp6: q6asm-dai: do not close port if its not opened sound/soc/qcom/qdsp6/q6afe-dai.c | 179 +++++++++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/q6afe.c | 21 +++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++ 4 files changed, 216 insertions(+), 16 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH - PCM rate v2 1/1] pcm: rate: Add error check for snd_pcm_avail_update()
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without these changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly. With this patch the returned error code of snd_pcm_avail_update() will be returned by snd_pcm_rate_avail_update(). Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Compared to v1[1] of this patch only the changelog and the subject has changed. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137838.html diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

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[alsa-devel] [PATCH 1/2] ALSA: pcm: trace XRUN event at injection, too
by Takashi Iwai 04 Jul '18

04 Jul '18

The PCM xrun injection triggers directly snd_pcm_stop() without the standard xrun handler, hence it's not recorded on the event buffer. Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl. They are inconvenient from the debugging POV. Let's make them to trigger XRUN via the standard helper more consistently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm.c | 2 +- sound/core/pcm_lib.c | 7 ++++--- sound/core/pcm_local.h | 2 ++ sound/core/pcm_native.c | 8 ++++---- 4 files changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c352bfb973cc..6f037a4b8b52 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -497,7 +497,7 @@ static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, snd_pcm_stream_lock_irq(substream); runtime = substream->runtime; if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irq(substream); } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..c1d2e8e1fc6b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -153,7 +153,8 @@ EXPORT_SYMBOL(snd_pcm_debug_name); dump_stack(); \ } while (0) -static void xrun(struct snd_pcm_substream *substream) +/* call with stream lock held */ +void __snd_pcm_xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +202,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, } } else { if (avail >= runtime->stop_threshold) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } } @@ -297,7 +298,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); + __snd_pcm_xrun(substream); return -EPIPE; } if (pos >= runtime->buffer_size) { diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 7a499d02df6c..c515612969a4 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -65,4 +65,6 @@ static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} #endif +void __snd_pcm_xrun(struct snd_pcm_substream *substream); + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cecc79772c94..20174d0c0527 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1337,13 +1337,12 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) { unsigned long flags; - int ret = 0; snd_pcm_stream_lock_irqsave(substream, flags); if (snd_pcm_running(substream)) - ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); snd_pcm_stream_unlock_irqrestore(substream, flags); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); @@ -1591,7 +1590,8 @@ static int snd_pcm_xrun(struct snd_pcm_substream *substream) result = 0; /* already there */ break; case SNDRV_PCM_STATE_RUNNING: - result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + __snd_pcm_xrun(substream); + result = 0; break; default: result = -EBADFD; -- 2.18.0

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[alsa-devel] [PATCH - PCM rate 1/1] pcm: rate: Use signed variable for return of snd_pcm_avail_update
by twischer＠de.adit-jv.com 04 Jul '18

04 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this changes a negative error code returned by snd_pcm_avail_update() will be not handled correctly Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 4e0c7ca..031a43d 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -970,9 +970,12 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) { snd_pcm_rate_t *rate = pcm->private_data; snd_pcm_t *slave = rate->gen.slave; - snd_pcm_uframes_t slave_size; + snd_pcm_sframes_t slave_size; slave_size = snd_pcm_avail_update(slave); + if (slave_size < 0) + return slave_size; + if (pcm->stream == SND_PCM_STREAM_CAPTURE) goto _capture; snd_pcm_rate_sync_hwptr(pcm); -- 2.7.4

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[alsa-devel] [PATCH v3 00/13] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 04 Jul '18

04 Jul '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 v3 is not split into two batches as done for v1 and v2, but keeps the same logical progression. The first three patches are mostly data structure changes, the DSP support capability is added then with an ASoC HDA driver and the last patches are fixes required for Skylake+. The changes to the HDAudio legacy driver are minimal. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). 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. 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. Rakesh Ughreja (13): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda: split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ALSA: hdac: ext: add wait for codec to respond after link reset ASoC: Intel: Skylake: fix widget handling include/sound/hdaudio.h | 29 ++ include/sound/hdaudio_ext.h | 123 ++----- sound/hda/ext/hdac_ext_bus.c | 80 ++--- sound/hda/ext/hdac_ext_controller.c | 64 ++-- sound/hda/ext/hdac_ext_stream.c | 104 +++--- sound/pci/hda/hda_bind.c | 12 + sound/pci/hda/hda_codec.c | 68 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 469 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/codecs/hdac_hdmi.c | 416 +++++++++++------------- sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 133 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 +++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 174 ++++++++++ sound/soc/intel/skylake/skl-messages.c | 50 ++- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 182 +++++++---- sound/soc/intel/skylake/skl-topology.c | 23 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 296 +++++++++++------ sound/soc/intel/skylake/skl.h | 13 +- 25 files changed, 1640 insertions(+), 691 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 -- 2.14.1

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[alsa-devel] [PATCH v4 0/7] soundwire: Add multi link support
by Shreyas NC 04 Jul '18

04 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters changes in v3: - Added comments to explain the trigger for multi link bank switch - Added comments in the intel_post_bank_switch ops to explain how synchronous trigger of bank switch is handled. Sanyog Kale (2): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (4): soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 483 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 467 insertions(+), 131 deletions(-) -- 2.7.4

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