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

[alsa-devel] [PATCH] ASoC: nau8824: use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 06 Jul '18

06 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: 1432039 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 6bd1445..468d514 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1274,7 +1274,7 @@ static int nau8824_calc_fll_param(unsigned int fll_in, 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] regmap: slimbus: add support to raw read/writes
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

SLIMbus supports upto 16 bytes in value management messages, so add support to raw read/writes upto 16 bytes. Also useful for paged register access on SLIMbus interfaced codecs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/base/regmap/regmap-slimbus.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap-slimbus.c b/drivers/base/regmap/regmap-slimbus.c index 91d501eda8a9..fc07518666b2 100644 --- a/drivers/base/regmap/regmap-slimbus.c +++ b/drivers/base/regmap/regmap-slimbus.c @@ -31,11 +31,30 @@ static int regmap_slimbus_byte_reg_write(void *context, unsigned int reg, return slim_writeb(sdev, reg, val); } +static int regmap_slimbus_write(void *context, const void *data, size_t count) +{ + struct slim_device *sdev = context; + + return slim_write(sdev, *(u16 *)data, count - 2, (u8 *)data + 2); +} + +static int regmap_slimbus_read(void *context, const void *reg, size_t reg_size, + void *val, size_t val_size) +{ + struct slim_device *sdev = context; + + return slim_read(sdev, *(u16 *)reg, val_size, val); +} + static struct regmap_bus regmap_slimbus_bus = { .reg_write = regmap_slimbus_byte_reg_write, .reg_read = regmap_slimbus_byte_reg_read, - .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, - .val_format_endian_default = REGMAP_ENDIAN_LITTLE, + .max_raw_write = 16, + .max_raw_read = 16, + .write = regmap_slimbus_write, + .read = regmap_slimbus_read, + .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, + .val_format_endian_default = REGMAP_ENDIAN_NATIVE, }; static const struct regmap_bus *regmap_get_slimbus(struct slim_device *slim, -- 2.16.2

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[alsa-devel] Can't detect "XRUNs"
by Raphaël Dingé 05 Jul '18

05 Jul '18

Hi, I don't get audio drop notification "XRUNs" in /proc/kmsg. Not sure if this is the right mailing list as maybe my question is more kernel related, but I'll try a shot here. I'm currently making a proof of concept to see if I can run a quite CPU hungry audio application running without audio drops over time given a set of tests. In particular I want to automate playback buffer underruns detection so that I don't have to sit and listen closely when they happen. With some very extreme tests, I can hear that there are audio drops all over the place. I have 2 audio cards in the system, one is audio over HDMI, and the other one is an USB audio card that supports audio class definition. In both case the sound is working as I would expect. The system itself is an Intel NUC i7. I configured the kernel with CONFIG_SND_DEBUG=y, CONFIG_SND_PCM_XRUN_DEBUG=y and CONFIG_SND_VERBOSE_PROCFS=y as well as echo 5 > /proc/asound/cardX/pcmYp/xrun_debug with X and Y the card and pcm stream of interest, and so hw:X,Y when opening the device with snd_pcm_open. Then I watch the output of the ksyslog interface using cat /proc/kmsg When an audio drop occurs, there is nothing about XRUNs or whatever it is called. I don't even know what kind of log message it should look like, and can't find it on google as well. With audio over HDMI, I have some occurence of ALSA: PCM: [Q] Lost interrupts? that seems more of less related to audio drops, but not always. With the USB audio card, I have nothing. I tried to write to the xrun_injection proc file in the stream, but nothing happens as well. Finally I log any occurence of snd_pcm_writei returning a negative number, but that never happens as well. Am I missing something completely? Thanks for any information on that matter, Raphael

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[alsa-devel] [PATCH v2 0/2] slimbus: Add Stream Support
by Srinivas Kandagatla 05 Jul '18

05 Jul '18

Thanks to Vinod and Stephen for reviewing v1 patchset This v2 patchset adds basic stream support for SLIMbus devices and controllers. Mostly inspired by soundwire stream patches. But slimbus stream is much simpler compared to soundwire >From slim_device side, we have below 6 new apis. slim_stream_allocate() - allocating runtime slim stream slim_stream_prepare() - to configure runtime stream with config slim_stream_enable() - enable channels/ports for data slim_stream_disable() - disable channels/ports. slim_stream_unprepare() - un configure runtime stream slim_stream_free() - free all associated memory. >From Controller side: Data channel Management and reconfiguration messages are applicable for all the controllers which are inline with SLIMbus specs. However light weight controller like NGD which have user specific implementation of some messages need to be invoked instead of standard message commands. For usecases like this simple enable/disable stream apis are provided. Assumptions: 1> Current design assumes that the channel and ports are statically allocated to the device during SoC integration, which is the case with all the Qualcomm SoCs. 2> One-to-One mapping between Port and Channel, SLIMBus versions earlier than 2.0 has only one endpoint per port. Current patchset can be extended to support more than one endpoints per port. 3> Only audio usecase, This patchset only supports Isochronous and Push/Pull transport protocols, which are sufficient for audio use cases. 4> DSP does all the data handling for the allocated channels. Which is true for Qcom SoCs. TODO: Bandwidth management. Dependency: This patchset has dependency on the NGD driver https://patchwork.kernel.org/patch/10474959/ Tested this patchset with WCD9335 codec playback and capture on DB820c using 4.18-rc1 with qdsp6 + slim tx patches merged in sound.git. I have pushed my working branch to [1] incase someone want to try. [1]:https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimb… Changes since v1 (https://lwn.net/Articles/758100/): - removed redundant state machine checks for audio usecases, suggested by Vinod. - fixed few typos. - removed irqsave/restore spinlock variant spotted by Vinod. - general cleanups suggested by Stephen and Vinod. thanks, srini Srinivas Kandagatla (2): slimbus: stream: add stream support slimbus: ngd: add stream support Documentation/driver-api/slimbus.rst | 5 + drivers/slimbus/Makefile | 2 +- drivers/slimbus/core.c | 2 + drivers/slimbus/qcom-ngd-ctrl.c | 149 ++++++++++- drivers/slimbus/slimbus.h | 189 ++++++++++++++ drivers/slimbus/stream.c | 477 +++++++++++++++++++++++++++++++++++ include/linux/slimbus.h | 46 ++++ 7 files changed, 867 insertions(+), 3 deletions(-) create mode 100644 drivers/slimbus/stream.c -- 2.16.2

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[alsa-devel] [PATCH - JACK IO plug v2 0/3] snd_pcm_drain support
by twischer＠de.adit-jv.com 05 Jul '18

05 Jul '18

Now in a clean manner as composed patches. pcm: ioplug: Provide avail helper function --------- No changes jack: Avoid call to snd_pcm_avail_update() from JACK thread --------- v2 uses snd_pcm_uframes_t as type for avail and does not check for avail < 0 becasue snd_pcm_ioplug_avail() will never retrun a value < 0 jack: Update poll_fd also in draining state --------- This patch is depending on [1]. Therefore it was rebased on top of [1]. Additionally the commit message was extended. v1 of this patch can be found in [2]. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137775.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-July/137769.html

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit
by Gustavo A. R. Silva 05 Jul '18

05 Jul '18

Add suffix ULL to constant 64 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: rate[index] * txclk_df * 64 Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9b59d87..740b90d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1118,7 +1118,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * txclk_df * 64; + rate_ideal = rate[index] * txclk_df * 64ULL; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- 2.7.4

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[alsa-devel] [PATCH 1/1] ASoC: rsnd: cmd: Add missing newline to debug message
by jiada_wang＠mentor.com 05 Jul '18

05 Jul '18

From: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> To comply with the style of all kernel messages, add newline to the end of every message. Fixes: 70fb10529f61 ("ASoC: rsnd: add MIX (Mixer) support") Signed-off-by: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/cmd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index d8043ad33540..cc191cd5fb82 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -86,7 +86,7 @@ static int rsnd_cmd_init(struct rsnd_mod *mod, cmd_case[rsnd_mod_id(src)] << 16; } - dev_dbg(dev, "ctu/mix path = 0x%08x", data); + dev_dbg(dev, "ctu/mix path = 0x%08x\n", data); rsnd_mod_write(mod, CMD_ROUTE_SLCT, data); rsnd_mod_write(mod, CMD_BUSIF_MODE, rsnd_get_busif_shift(io, mod) | 1); -- 2.17.0

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[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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