March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/12 v2] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 19 Mar '15

19 Mar '15

Hi Mark These are v2 of DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : new Renesas sound card driver 5 : tidyup for ak4642 6 - 10 : fixup/tidyup patches for rsnd 11 - 12 : DPCM support patch for rsnd Kuninori Morimoto (12): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 4) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 5) ASoC: ak4642: tidyup DAPM route for playback 6) ASoC: rsnd: no more SSI restart when unusual situation 7) ASoC: rsnd: no more SRC restart when unusual situation 8) ASoC: rsnd: tidyup error message format 9) ASoC: rsnd: show debug info for sampling rate convert 10) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 11) ASoC: rsnd: add dai_link stream name 12) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/codecs/ak4642.c | 4 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 496 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ 13 files changed, 765 insertions(+), 66 deletions(-) Best regards --- Kuninori Morimoto _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] bug in alsa-plugins: #if SND_PCM_EXTPLUG_VERSION wrong
by Ethan Grammatikidis 18 Mar '15

18 Mar '15

in alsa-plugins 1.0.28 and 1.0.29, mix/*.c contain the following line: #if SND_PCM_EXTPLUG_VERSION >= 0x102 that 0x102 needs to be 0x10002. i have alsa-lib 1.0.26 installed. its headers define SND_PCM_EXTPLUG_VERSION to 0x10001, corresponding to version number 1.0.1. that #if effectively tests for >= 0.1.2, so attempting to build alsa-plugins 1.0.28 or 1.0.29 results in a long list of errors. SND_CHMAP_MONO, SND_CHMAP_FL, etc are undeclared and snd_pcm_chmap_query_t is an unknown type. in alsa-plugins 1.0.26 SND_PCM_EXTPLUG_VERSION is not tested. i didn't look at 1.0.27 reporting here because the bug tracker is down. -- Wir mussen wissen, wir werden wissen.

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[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..f65ac19 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: Intel: Remove support for Intel MID DMA from firmware loader
by Jarkko Nikula 18 Mar '15

18 Mar '15

Intel MID DMA driver is going to be removed by the coming commit 36111da7838e ("dmaengine: intel-mid-dma: remove the driver") in spi.git tree. Since there are no users for SST_DMA_TYPE_MID type the support for it can be removed from here in advance. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-dsp.h | 1 - sound/soc/intel/sst-firmware.c | 3 --- 2 files changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32f0077..3412474083ff 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -28,7 +28,6 @@ /* Supported SST DMA Devices */ #define SST_DMA_TYPE_DW 1 -#define SST_DMA_TYPE_MID 2 /* autosuspend delay 5s*/ #define SST_RUNTIME_SUSPEND_DELAY (5 * 1000) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5e5800897da2..38881f1fb990 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -277,9 +277,6 @@ int sst_dma_new(struct sst_dsp *sst) case SST_DMA_TYPE_DW: dma_dev_name = "dw_dmac"; break; - case SST_DMA_TYPE_MID: - dma_dev_name = "Intel MID DMA"; - break; default: dev_err(sst->dev, "error: invalid DMA engine %d\n", sst->pdata->dma_engine); -- 2.1.4

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[alsa-devel] [PATCH] ASoC: samsung: Enable SND_SIMPLE_CARD for Odroid X2/U3
by Sylwester Nawrocki 18 Mar '15

18 Mar '15

Odroid X2/U3 sound support can now be specified in device tree using the simple card binding, make SND_SOC_ODROIDX2 select SND_SIMPLE_CARD to ensure there are always required drivers in place. Reported-by: Tobias Jakobi <liquid.acid(a)gmx.net> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 0632a36..25caadf 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -226,6 +226,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroid-X2/U3. -- 1.7.9.5

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Re: [alsa-devel] ASoC: Intel: mrfld- add ACPI module
by Dan Carpenter 18 Mar '15

18 Mar '15

Hello Vinod Koul, The patch 336cfbb05edf: "ASoC: Intel: mrfld- add ACPI module" from Nov 11, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_acpi.c:313 sst_acpi_probe() info: ignoring unreachable code. sound/soc/intel/sst/sst_acpi.c 304 ret = sst_context_init(ctx); 305 if (ret < 0) 306 return ret; 307 308 /* need to save shim registers in BYT */ 309 ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), 310 GFP_KERNEL); 311 if (!ctx->shim_regs64) { 312 return -ENOMEM; ^^^^^^^^^^^^^^^ 313 goto do_sst_cleanup; ^^^^^^^^^^^^^^^^^^^^ 314 } 315 316 sst_configure_runtime_pm(ctx); 317 platform_set_drvdata(pdev, ctx); 318 return ret; 319 320 do_sst_cleanup: 321 sst_context_cleanup(ctx); 322 platform_set_drvdata(pdev, NULL); 323 dev_err(ctx->dev, "failed with %d\n", ret); 324 return ret; 325 } regards, dan carpenter

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[alsa-devel] [GIT PULL] ASoC fixes for v4.0
by Mark Brown 18 Mar '15

18 Mar '15

The following changes since commit 06e5801b8cb3fc057d88cb4dc03c0b64b2744cda: Linux 4.0-rc4 (2015-03-15 17:38:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.0-rc4 for you to fetch changes up to 33484c67398295108ab31e17b50b5a6fba1f5eef: Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus (2015-03-16 12:03:17 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.0 As well as the usual collection of driver specific fixes there's a few more generic things: - Lots of fixes from Takashi for drivers using the wrong field in the control union to communicate with userspace, leading to potential errors on 64 bit systems. - A fix from Lars for locking of the lists of devices we maintain, mostly only likely to trigger during device probe and removal. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt286: Change the DMI mapping for Dino Eric Nelson (1): ASoC: sgtl5000: remove useless register write clearing CHRGPUMP_POWERUP Lars-Peter Clausen (4): ASoC: ak4671: Fix control-less DAPM routes ASoC: da732x: Fix control-less DAPM routes ASoC: sn95031: Fix control-less DAPM routes ASoC: Fix component lists locking Lu, Han (1): ASoC: Intel: remove conflicts when load/unload multiple firmware images Maciej S. Szmigiero (1): fsl_ssi: fix of_property_read_u32_array return value check Mark Brown (3): Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4671', 'asoc/fix/control', 'asoc/fix/da732x', 'asoc/fix/fsl-ssi', 'asoc/fix/lock' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000' and 'asoc/fix/sn95031' into asoc-linus Nicolin Chen (1): ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation Takashi Iwai (14): ASoC: adav80x: Fix wrong value references for boolean kctl ASoC: ak4641: Fix wrong value references for boolean kctl ASoC: cs4271: Fix wrong value references for boolean kctl ASoC: es8238: Fix wrong value references for boolean kctl ASoC: pcm1681: Fix wrong value references for boolean kctl ASoC: tas5086: Fix wrong value references for boolean kctl ASoC: wm2000: Fix wrong value references for boolean kctl ASoC: wm8731: Fix wrong value references for boolean kctl ASoC: wm8903: Fix wrong value references for boolean kctl ASoC: wm8904: Fix wrong value references for boolean kctl ASoC: wm8955: Fix wrong value references for boolean kctl ASoC: wm8960: Fix wrong value references for boolean kctl ASoC: wm9712: Fix wrong value references for boolean kctl ASoC: wm9713: Fix wrong value references for boolean kctl sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/ak4671.c | 44 +++++++++++++++++++-------------------- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/pcm1681.c | 4 ++-- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/sgtl5000.c | 8 +------ sound/soc/codecs/sn95031.c | 4 ++-- sound/soc/codecs/tas5086.c | 4 ++-- sound/soc/codecs/wm2000.c | 8 +++---- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/intel/sst-haswell-dsp.c | 3 --- sound/soc/intel/sst-haswell-ipc.c | 32 +++++++++++++++++++++------- sound/soc/soc-core.c | 41 ++++++++++++++++++++++++++---------- 23 files changed, 118 insertions(+), 92 deletions(-)

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 18 Mar '15

18 Mar '15

Set a rule constraints to allow only combinations of sample-rate and number of frame-bits that can be played/captured with a reasonable sample-rate accuracy. Also, takes the number of serializers into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Now this works as it should. Thanks, Jyri sound/soc/davinci/davinci-mcasp.c | 150 +++++++++++++++++++++++++++++++++++--- 1 file changed, 140 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3ecee3b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -64,6 +65,11 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -98,6 +104,8 @@ struct davinci_mcasp { #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif + + struct davinci_mcasp_ruledata ruledata[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -855,6 +863,30 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, return 0; } +static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, + unsigned int bclk_freq, + int *error_ppm) +{ + int div = mcasp->sysclk_freq / bclk_freq; + int rem = mcasp->sysclk_freq % bclk_freq; + + if (rem != 0) { + if (div == 0 || + ((mcasp->sysclk_freq / div) - bclk_freq) > + (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + div++; + rem = rem - bclk_freq; + } + } + if (error_ppm) + *error_ppm = + (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) + /div - 1000000; + + return div; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -872,16 +904,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - unsigned int bclk_freq = snd_soc_params_to_bclk(params); - unsigned int div = mcasp->sysclk_freq / bclk_freq; - if (mcasp->sysclk_freq % bclk_freq != 0) { - if (((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) - div++; - dev_warn(mcasp->dev, - "Inaccurate BCLK: %u Hz / %u != %u Hz\n", - mcasp->sysclk_freq, div, bclk_freq); - } + uint bclk_freq = snd_soc_params_to_bclk(params); + int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + TX_MODE : RX_MODE; + int ppm, div; + + bclk_freq /= mcasp->ruledata[dir].serializers; + + div = davinci_mcasp_calc_clk_div(mcasp, bclk_freq, &ppm); + + if (ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + ppm); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } @@ -973,6 +1008,74 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static const unsigned int davinci_mcasp_dai_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, +}; + +#define DAVINCI_MAX_RATE_ERROR_PPM 1000 + +static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int frame_size; + int i, count = 0; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = frame_size*davinci_mcasp_dai_rates[i]/ + rd->serializers; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = davinci_mcasp_dai_rates[i]; + + } + dev_dbg(rd->mcasp->dev, "%d frequencies for %u fsize (%d)\n", + count, frame_size, rd->serializers); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_framebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_interval *ci = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + unsigned int max_chan_per_wire = ci->max / rd->serializers; + unsigned int list[max_chan_per_wire*4]; + int rate = params_rate(params); + int chans, fbits, count = 0; + + for (chans = ci->min; chans <= max_chan_per_wire; chans++) { + for (fbits = 8; fbits <= 32; fbits += 8) { + uint bclk_freq = chans*fbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + list[count++] = chans*fbits*rd->serializers; + BUG_ON(count > ARRAY_SIZE(list)); + } + } + } + + dev_dbg(rd->mcasp->dev, + "%d possible frame size for %d Hz (%d)\n", + count, rate, rd->serializers); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -998,6 +1101,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } + mcasp->ruledata[dir].serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1012,6 +1116,32 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + + /* + * If we rely on implicit BCLK divider setting we should + * set constraints based on what we can provide. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { + int ret; + + mcasp->ruledata[dir].mcasp = mcasp; + + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + davinci_mcasp_hw_rule_rate, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + davinci_mcasp_hw_rule_framebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] Support two wm8962 codec in i.mx6 board.
by Shawn Embedded 18 Mar '15

18 Mar '15

Hello everyone, I'm new to ALSA. I have a question here. Now there is a wm8962 on the i.mx6 board, running well, and we want to add an extra wm8962 to the board, connected to i.mx6 on another SSI port through a different i.mx6's AUDMUX port. The two wm8962 will work independently completely, at the same time possiblly. Before sending this mail to the list, I've already googled some materials about my requirements. I've found there is a structure named "struct snd_soc_aux_dev". For my applicantion, if I want to support another wm8962, could I can use this structure for my intention please? But in this link http://www.spinics.net/lists/alsa-devel/msg40127.html , it's said that This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. then, does it mean that I couldn't use this structure for my application? Or, I should declare another structure "struct snd_soc_card" for the second wm8962, just as the now-working wm8962? or other ways? In fact, I'm a little confused if I could declare two structure snd_soc_card at the same? If it's OK, what should I do next to meet my requirement? Could anyone give me any guidance on this? Any links, and materials will also be much appreciated. Thank you in advance. ------------------- Best Regards, Shawn

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[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 18 Mar '15

18 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6d3412f..9906119 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

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