June 2016 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: rt5668: add rt5668 codec driver
by Mark Brown 20 Jun '16

20 Jun '16

On Thu, Jun 16, 2016 at 11:31:07AM +0000, Jack Yu wrote: > [Mark] Given that the interrupt handler just schedules a work item there's no > need for it to be a threaded interrupt, it can just be a normal > interrupt and save the bother of masking the hardware interrupt and > running a thread. > Do you mean that it's not appropriate to use " queue_delayed_work" here, we should > use other method for handling work item ? No, when I said it shouldn't be a threaded interrupt I meant that using request_threaded_irq() doesn't seem appropriate.

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[alsa-devel] [PATCH] ASoC: core: Remove selects of regmap buses
by Mark Brown 20 Jun '16

20 Jun '16

We used to have code wrapping the regmap I2C and SPI buses as part of the conversion of ASoC drivers from using the ASoC level I/O to the regmap level I/O. Since all drivers have now been converted and the core code removed we can remove the per-bus selects and instead just select the regmap core. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 182d92efc7c8..4935e23ff3d6 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,8 +7,7 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK - select REGMAP_I2C if I2C - select REGMAP_SPI if SPI_MASTER + select REGMAP ---help--- If you want ASoC support, you should say Y here and also to the -- 2.8.1

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[alsa-devel] [GIT PULL]: Skylake audio firmware update
by Vinod Koul 20 Jun '16

20 Jun '16

Hi, Please PULL to receive the latest Skylake audio firmware. The following changes since commit a4bbc8112cc04fcc0e0861e8b72197b6cb83ab83: linux-firmware: Update firmware file for Intel Bluetooth 8260 (2016-06-09 08:11:22 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 80aae2270140807c4d8e304c39def1fe9f93edb9: linux-firmware: intel: Update Skylake audio firmware (2016-06-17 14:01:52 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v948.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v948.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] ASoC: Intel: Add support for PM ops in bxt-da7219_max98357a
by Vinod Koul 20 Jun '16

20 Jun '16

We need card to be early suspended and late resumed, so use prepare and complete for card suspend and resume. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b117d365..df5f269a9b67 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -441,11 +441,30 @@ static int broxton_audio_probe(struct platform_device *pdev) return devm_snd_soc_register_card(&pdev->dev, &broxton_audio_card); } +#ifdef CONFIG_PM_SLEEP + +static void broxton_complete(struct device *dev) +{ + snd_soc_resume(dev); +} + +#else +#define broxton_complete NULL +#endif + +static const struct dev_pm_ops broxton_pm_ops = { + .prepare = snd_soc_suspend, + .complete = broxton_complete, + .freeze = snd_soc_suspend, + .thaw = snd_soc_resume, + .poweroff = snd_soc_poweroff, + .restore = snd_soc_resume, +}; static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { .name = "bxt_da7219_max98357a_i2s", - .pm = &snd_soc_pm_ops, + .pm = &broxton_pm_ops, }, }; module_platform_driver(broxton_audio) -- 1.9.1

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[alsa-devel] [PATCH v1 1/2] ASoC: Add DT bindings documentation for max98504 amplifier
by Sylwester Nawrocki 20 Jun '16

20 Jun '16

This patch adds DT bindings documentation for Maxim MAX98504 speaker amplifier. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since initial version: - dropped all optional properties used for setting up TDM and speaker data source, - added regulator supply properties. --- .../devicetree/bindings/sound/max98504.txt | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt new file mode 100644 index 0000000..7aa5b93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98504.txt @@ -0,0 +1,22 @@ +Maxim MAX98504 class D mono speaker amplifier + +This device supports I2C control interface and an IRQ output signal. It features +a PCM and PDM digital audio interface (DAI) and a differential analog input. + +Required properties: + + - compatible : "maxim,max98504" + - reg : should contain the I2C slave device address + - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + + max98504: max98504@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DVDD-supply = <&regulator>; + DIOVDD-supply = <&regulator>; + PVDD-supply = <&regulator>; +}; -- 1.9.1

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[alsa-devel] [PATCH] alsa-tools gcc6 narrowing error
by Michel Normand 20 Jun '16

20 Jun '16

Trying to build alsa-tools with gcc6 for PowerPC architecture. It failed as detailed in proposed patch of next mail Michel Normand (1): gcc6 narrowing error hdspmixer/src/channelmap.cxx | 158 +++++++++++++++++++++---------------------- us428control/Cus428Midi.cc | 30 ++++---- 2 files changed, 94 insertions(+), 94 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 2/2] ASoc: wm8731: add normal mode with 12MHz XTAL
by Matt Flax 20 Jun '16

20 Jun '16

I have realised that this second patch may not be necessary. It is also likely that it will skew the actual sample rate. Charles, can you please advise me on the following... I notice that I can probably get the correct sample rates by spoofing the crystal frequencies in my machine driver. For example ... The ratio between the USB and Normal mode fs multipliers with BOSR=0 is : 250/256 = 0.97656 Similarly, I can match this with the following crystal frequency ratio (crystal f in MHz) : 12/12.288 = 0.97656 By induction, I assume that I can spoof a 12.288 clock with BOSR=0 for sample rate = 48kHz, 96 kHz when using a 12 MHz crystal. Again, I can match ratios between the following crystal frequecies (in MHz) : 12/16.9344 = 0.70862 With BOSR = 1, I assume that can get 44.1 kHz and 88.2 kHz because the fs ratio between normal and usb mode is : 272/384 = 0.70833 Now these two ratios are skew by 0.41% which is equvalent to the skew in 12 MHz USB mode clocks of 44.1kHz to 44.118kHz as reported in the wm8731 datasheet. Can you advise me whether my idea of spoofing clocks in the machine driver is perhaps a better way of acquiring the correct sample rates, rather then this patch ? thanks Matt On 10/06/16 10:11, Matt Flax wrote: > This patch allows a machine to set normal mode when using a > 12 MHz crystal. In USB_MODE the 12 MHz crystal signal is output on > the BCLK pin. In NORMAL_MODE the bit clock is output on the BCLK pin. > The previously ignored direction variable input to the > wm8731_set_dai_sysclk function is used to indicate whether crystal > or bit clock is output on the BCLK codec pin. > I have ensured that this does not effect the db1200.c machine driver > (the only other driver to use a 12 MHz crystal in USB_MODE). It also > does not effect the other machine drivers which use the wm8731 codec > as they don't use USB_MODE nor 12 MHz crystals. > > When driving the wm8731 in normal mode with a 12 MHz crystal, the > DSP filters are expected to be effected. If the filters are activated > then it is expected that the filter cut offs are shifted by 2.3% at > 250 fs and 29.1% at 272 fs. The assumption is that the filters are > linear, which implies a linear shift in rate scaling effects the > filter cut off frequencies linearly. If the filters are not active, > there will be no effects. > > Signed-off-by: Matt Flax <flatmax(a)flatmax.org> > --- > sound/soc/codecs/wm8731.c | 14 ++++++++++++-- > 1 file changed, 12 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c > index d18261a..ec98d5a 100644 > --- a/sound/soc/codecs/wm8731.c > +++ b/sound/soc/codecs/wm8731.c > @@ -5,6 +5,7 @@ > * Copyright 2006-12 Wolfson Microelectronics, plc > * > * Author: Richard Purdie <richard(a)openedhand.com> > + * Author: Matt Flax <flatmax(a)flatmax.com> 12Mhz XTAL normal mode and 32bit mode. > * > * Based on wm8753.c by Liam Girdwood > * > @@ -53,6 +54,7 @@ struct wm8731_priv { > int sysclk_type; > int playback_fs; > bool deemph; > + bool bclk_12_mhz; > > struct mutex lock; > }; > @@ -341,8 +343,11 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, > struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); > u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; > int i = get_coeff(wm8731->sysclk, params_rate(params)); > - u16 srate = (coeff_div[i].sr << 2) | > - (coeff_div[i].bosr << 1) | coeff_div[i].usb; > + u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1); > + > + /* this determines whether to output bit or crystal clk */ > + if (wm8731->bclk_12_mhz) > + srate |= coeff_div[i].usb; > > wm8731->playback_fs = params_rate(params); > > @@ -420,6 +425,11 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, > > wm8731->sysclk = freq; > > + if (dir == SND_SOC_CLOCK_IN) /* output the 12 MHz clock */ > + wm8731->bclk_12_mhz = 1; > + else /* SND_SOC_CLOCK_OUT, output the bit clock */ > + wm8731->bclk_12_mhz = 0; > + > snd_soc_dapm_sync(dapm); > > return 0;

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[alsa-devel] [PATCH] ASoC: compress: Pass error out of soc_compr_pointer
by Charles Keepax 20 Jun '16

20 Jun '16

Both soc_compr_pointer and the platform driver pointer callback return ints but current soc_compr_pointer always returns 0. Update this so we return the actual value from the platform driver callback. This doesn't fix any issues simply makes the code more consistent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 875733c..d2df46c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) - platform->driver->compr_ops->pointer(cstream, tstamp); + ret = platform->driver->compr_ops->pointer(cstream, tstamp); mutex_unlock(&rtd->pcm_mutex); - return 0; + return ret; } static int soc_compr_copy(struct snd_compr_stream *cstream, -- 2.1.4

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[alsa-devel] [PATCH 1/1] ASoC: pcm5102a: Support 352k8 and 384k sample rates
by Clive Messer 19 Jun '16

19 Jun '16

Add support for 352k8 and 384k samples rates to pcm5102a codec driver, with SNDRV_PCM_RATE_KNOT and a constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm5102a.c | 38 +++++++++++++++++++++++++++++++++++++- 1 file changed, 37 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index ed51567..53dfc99 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -20,16 +20,52 @@ #include <sound/soc.h> +static const u32 pcm5102a_rates[] = { + 8000, 16000, 32000, 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000, +}; + +static const struct snd_pcm_hw_constraint_list pcm5102a_constraint_rates = { + .count = ARRAY_SIZE(pcm5102a_rates), + .list = pcm5102a_rates, +}; + +static int pcm5102a_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm5102a_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "%s: set rates (8k-384k) constraint\n", __func__); + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &pcm5102a_constraint_rates); + if (ret != 0) { + dev_err(codec->dev, "%s: Failed to set rates constraint: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dai_ops pcm5102a_dai_ops = { + .startup = pcm5102a_dai_startup, +}; + static struct snd_soc_dai_driver pcm5102a_dai = { .name = "pcm5102a-hifi", .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, + .ops = &pcm5102a_dai_ops, }; static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; -- 2.5.5

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[alsa-devel] [PATCH 1/1] ASoC: pcm512x: Add 352k8 sample rate support
by Clive Messer 19 Jun '16

19 Jun '16

Add 352k8 sample rate to the slave constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 047c489..eca6648 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -480,7 +480,7 @@ static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, - 88200, 96000, 176400, 192000, 384000, + 88200, 96000, 176400, 192000, 352800, 384000, }; static const struct snd_pcm_hw_constraint_list constraints_slave = { -- 2.5.5

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