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

[alsa-devel] Can iMX6SL supports different sampling rate for different SSIs?
by Xuebing Wang 20 Mar '15

20 Mar '15

Nicolin, Thank you very much for your always help. I am using 3 SSIs (as I2S master mode) on iMX6SL for 3 different codec. 1) If I read iMX6SL reference manual correctly, I am deriving SSI sys_clock from pll4. -- pll4 = Fref * (DIV_SELECT + NUM/DENOM) ** I am using a value of 786.432000 MHz -- pll4_main_clk = pll4 / (PLL_AUDIOn: POST_DIV_SELECT) / CCM_ANALOG_MISC2n: MSB:LSB ** I am using a value of 196.608 MHz -- SSI1_CLK_ROOT = pll4_main_clk / (CS1CDR: ssi1_clk_pred) / CS1CDR: ssi1_clk_podf ** I am using a value of 3.072 MHz (for 48k sampling rate) 2) As pll4_main_clk is global for all 3 SSIs, I think that we can *not* use 8k for ssi1 and 44.1k for ssi2, right? Because 8k and 44.1k requires different pll4 clock, right? -- However, as 48k, 32k, 16k, 8k can use same pll4, thus different SSI can use different sampling rate in this subset (48k, 32k, 16k, 8k). -- Same principle applies for 44.1k, 22.050, 11.025 subset of sampling rates. Is my understanding correct? -- Xuebing

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[alsa-devel] [PATCH v2.1] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- This is othetwise the same patch as v2, but I added Peter's ack and rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/davinci sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d40b392..625cca1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> #include <linux/platform_data/davinci_asp.h> +#include <linux/math64.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -65,6 +66,11 @@ struct davinci_mcasp_context { bool pm_state; }; +struct davinci_mcasp_ruledata { + struct davinci_mcasp *mcasp; + int serializers; +}; + struct davinci_mcasp { struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; @@ -99,6 +105,8 @@ 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, @@ -868,6 +876,30 @@ 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) @@ -883,16 +915,20 @@ 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); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &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); } @@ -974,6 +1010,108 @@ 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; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + 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 %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(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); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + 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) { @@ -999,6 +1137,7 @@ 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 @@ -1013,6 +1152,42 @@ 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_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: remove misleading DMA error messages on Baytrail platforms
by Pierre-Louis Bossart 19 Mar '15

19 Mar '15

During probe, the Baytrail audio driver reports errors such as: [44.172040] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [44.172137] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed Those error messages are misleading, there is no error since the DMA is explicitly not configured for Baytrail. Add a test to remove DMA error checks when DMA is not configured and return silently. Acked-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/sst-acpi.c | 1 + sound/soc/intel/sst-dsp.h | 1 + sound/soc/intel/sst-firmware.c | 4 ++++ 3 files changed, 6 insertions(+) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index b3d8456..42f293f 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -142,6 +142,7 @@ static int sst_acpi_probe(struct platform_device *pdev) sst_acpi->desc = desc; sst_acpi->mach = mach; + sst_pdata->resindex_dma_base = desc->resindex_dma_base; if (desc->resindex_dma_base >= 0) { sst_pdata->dma_engine = desc->dma_engine; sst_pdata->dma_base = desc->resindex_dma_base; diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index f291e32..148d8c5 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -206,6 +206,7 @@ struct sst_pdata { const struct firmware *fw; /* DMA */ + int resindex_dma_base; /* other fields invalid if equals to -1 */ u32 dma_base; u32 dma_size; int dma_engine; diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 5f71ef6..4a993d1 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -271,6 +271,10 @@ int sst_dma_new(struct sst_dsp *sst) const char *dma_dev_name; int ret = 0; + if (sst->pdata->resindex_dma_base == -1) + /* DMA is not used, return and squelsh error messages */ + return 0; + /* configure the correct platform data for whatever DMA engine * is attached to the ADSP IP. */ switch (sst->pdata->dma_engine) { -- 1.9.1

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[alsa-devel] [PATCH] ALSA: jack: create jack kcontrols for every jack input device
by Jie Yang 19 Mar '15

19 Mar '15

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Currently the ALSA jack core registers only input devices for each jack registered. These jack input devices are not readable by userspace devices that run as non root. This patch adds support for additionally registering jack kcontrol devices for every input jack registered. This allows non root userspace to read jack status. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Modified-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reveiwed-by: Mark Brown <broonie(a)kernel.org> --- include/sound/control.h | 2 ++ include/sound/jack.h | 2 ++ sound/core/jack.c | 89 +++++++++++++++++++++++++++++++++++++++++++++++-- 3 files changed, 91 insertions(+), 2 deletions(-) mode change 100644 => 100755 sound/core/jack.c diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..023b70a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -66,6 +66,7 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ + struct list_head jack_list; /* list of controls belong to the same jack*/ struct snd_ctl_elem_id id; unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; @@ -75,6 +76,7 @@ struct snd_kcontrol { snd_kcontrol_tlv_rw_t *c; const unsigned int *p; } tlv; + unsigned int jack_bit_idx; /*the corresponding jack type bit index */ unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); diff --git a/include/sound/jack.h b/include/sound/jack.h index 2182350..477dcb4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -73,6 +73,8 @@ enum snd_jack_types { struct snd_jack { struct input_dev *input_dev; + struct list_head kctl_list; + struct snd_card *card; int registered; int type; const char *id; diff --git a/sound/core/jack.c b/sound/core/jack.c old mode 100644 new mode 100755 index 8658578..c067b6a --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -24,6 +24,7 @@ #include <linux/module.h> #include <sound/jack.h> #include <sound/core.h> +#include <sound/control.h> static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, @@ -54,7 +55,13 @@ static int snd_jack_dev_disconnect(struct snd_device *device) static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + struct snd_kcontrol *kctl; + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + list_del(&kctl->jack_list); + snd_ctl_remove(card, kctl); + } if (jack->private_free) jack->private_free(jack); @@ -100,6 +107,63 @@ static int snd_jack_dev_register(struct snd_device *device) return err; } + +/* get the first unused/available index number for the given kctl name */ +static int get_available_index(struct snd_card *card, const char *name) +{ + struct snd_kcontrol *kctl; + int idx = 0; + int len = strlen(name); + + down_write(&card->controls_rwsem); + next: + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(name, kctl->id.name, len) && + !strcmp(" Jack", kctl->id.name + len) && + kctl->id.index == idx) { + idx++; + goto next; + } + } + up_write(&card->controls_rwsem); + return idx; +} + +static void kctl_private_free(struct snd_kcontrol *kctl) +{ + list_del(&kctl->jack_list); +} + +static int snd_jack_new_kctl(struct snd_card *card, struct snd_jack *jack, int type) +{ + struct snd_kcontrol *kctl; + int i, err, index, state = 0 /* use 0 for default state ?? */; + + INIT_LIST_HEAD(&jack->kctl_list); + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (type & testbit) { + index = get_available_index(card,jack->id); + kctl = snd_kctl_jack_new(jack->id, index, card); + if (!kctl) + return -ENOMEM; + kctl->private_free = kctl_private_free; + /* use jack_bit_idx for the kctl type bit */ + kctl->jack_bit_idx = i; + + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + list_add_tail(&kctl->jack_list, &jack->kctl_list); + + /* set initial jack state */ + snd_kctl_jack_report(card, kctl, state & testbit); + } + } + + return 0; +} + /** * snd_jack_new - Create a new jack * @card: the card instance @@ -138,7 +202,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, } jack->input_dev->phys = "ALSA"; - + jack->card = card; jack->type = type; for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) @@ -150,10 +214,17 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (err < 0) goto fail_input; + /* register jack kcontrols */ + err = snd_jack_new_kctl(card, jack, type); + if (err < 0) + goto fail_kctl; + *jjack = jack; return 0; +fail_kctl: + snd_device_free(card, jack); fail_input: input_free_device(jack->input_dev); kfree(jack->id); @@ -230,6 +301,7 @@ EXPORT_SYMBOL(snd_jack_set_key); */ void snd_jack_report(struct snd_jack *jack, int status) { + struct snd_kcontrol *kctl; int i; if (!jack) @@ -245,13 +317,26 @@ void snd_jack_report(struct snd_jack *jack, int status) for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) { int testbit = 1 << i; - if (jack->type & testbit) + if (jack->type & testbit) { input_report_switch(jack->input_dev, jack_switch_types[i], status & testbit); + } } input_sync(jack->input_dev); + + for (i = 0; i < fls(SND_JACK_BTN_0); i++) { + int testbit = 1 << i; + if (jack->type & testbit) { + list_for_each_entry(kctl, &jack->kctl_list, jack_list) { + if (kctl->jack_bit_idx == i) { + snd_kctl_jack_report(jack->card, kctl, + status & testbit); + } + } + } + } } EXPORT_SYMBOL(snd_jack_report); -- 1.9.1

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Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Peter Ujfalusi 19 Mar '15

19 Mar '15

On 03/19/2015 07:07 AM, Greg Knight wrote: > Hi, Mark, Jaroslav, > > I've attached a couple patches to provide support for the Freescale > SGTL5000 chip when using the McASP on the AM3359. It is more correct to say that this is needed in a custom hardware which uses McASP and SGTL5000 codec. > This adds an optional hack, "clock-to-talk" to the McASP subsystem which > can start AHCLKX/AHCLKR prior to initializing the codec. This is needed > on the SGTL5000, which depends on AHCLKX for its master clock. SO is it so that the codec's MCLK is coming from McASP AHCLKX (or R) and this clock need to be present all the time? W/o the MCLK the registers are not accessible? > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? First off, please send the patches with git send-email so people can comment on them. This will not work in mainline kernel. The McASP is turned off when it is not in use (no audio activity) so the AHCLKX/R is not going to be there on the ball output. How do you configure the frequency of the AHCLKX/R, this hack only enables it, so you are going to have _some_ clock going out. You should be using set_clkdiv and set_sysclk calls from the machine driver to configure the AHCLKX (we do not have AHCLKR supported in the driver ATM). -- Péter

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Re: [alsa-devel] Patches to bind the SGTL5000 chip to AM33XX McASP
by Mark Brown 19 Mar '15

19 Mar '15

On Thu, Mar 19, 2015 at 01:07:39AM -0400, Greg Knight wrote: > These patches are based off of 3.14.31-ti-r49. What is the process for > getting these merged? Please submit patches using the process in SubmittingPatches, you need to work against current development versions of the kernel rather than an old vendor kernel.

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[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Jyri Sarha 19 Mar '15

19 Mar '15

Set rule constraints to allow only combinations of sample-rate, sample-bits, and channels that can be played/captured with reasonable sample-rate accuracy. The logic with tdm-slots and serializers (=i2s data wires) goes like this: The first wire will take all channels up to number of tdm-slots, before following wires (if any) are used. If the first wire is used fully, the remaining wires share the same clocks and the divider can be calculated for the first wire. Also, takes the number of tdm-slots into account when implicitly selecting the BLCK divider. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I hope the logic with tdm-slots and serializers is now correct. I do not have HW to test multiple serializers case, so please review davinci_mcasp_hw_rule_channels() extra carefully. Best regards, Jyri sound/soc/davinci/davinci-mcasp.c | 195 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 185 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..3b2fb56 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 +904,20 @@ 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); - } + int channels = params_channels(params); + int rate = params_rate(params); + int sbits = params_width(params); + int ppm, div; + + if (channels > mcasp->tdm_slots) + channels = mcasp->tdm_slots; + + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + &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 +1009,108 @@ 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; + int sbits = params_width(params); + int channels = params_channels(params); + unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; + int i, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { + uint bclk_freq = sbits*channels*davinci_mcasp_dai_rates[i]; + 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 %d sbits and %d channels\n", + count, sbits, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_samplebits(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + int channels = params_channels(params); + int rate = params_rate(params); + unsigned int list[4]; + int sbits, count = 0; + + if (channels > rd->mcasp->tdm_slots) + channels = rd->mcasp->tdm_slots; + + for (sbits = 8; sbits <= 32; sbits += 8) { + uint bclk_freq = channels*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) + list[count++] = sbits; + } + dev_dbg(rd->mcasp->dev, + "%d possible sample size for %d Hz and %d channels\n", + count, rate, channels); + + return snd_interval_list(hw_param_interval(params, rule->var), + count, list, 0); +} + +static int davinci_mcasp_hw_rule_channels(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); + int sbits = params_width(params); + int rate = params_rate(params); + int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? + rd->mcasp->tdm_slots : ci->max; + unsigned int list[ci->max - ci->min + 1]; + int c1, c, count = 0; + + for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { + uint bclk_freq = c1*sbits*rate; + int ppm; + + davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + /* If we can use all tdm_slots, we can put any + amount of channels to remaining wires as + long as they fit in. */ + if (c1 == rd->mcasp->tdm_slots) { + for (c = c1; c <= rd->serializers*c1 && + c <= ci->max; c++) + list[count++] = c; + } else { + list[count++] = c1; + } + } + } + dev_dbg(rd->mcasp->dev, + "%d possible channel counts for %d Hz and %d sbits\n", + count, rate, sbits); + + 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 +1136,7 @@ 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 +1151,42 @@ 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_SAMPLE_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + davinci_mcasp_hw_rule_samplebits, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) + return ret; + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + davinci_mcasp_hw_rule_channels, + &mcasp->ruledata[dir], + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + if (ret) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Daniel 19 Mar '15

19 Mar '15

Hello, First of all, since this is my first message I hope am not mistaken reporting this two issues about this piece of hardware (I was pointed here by alsa irc support channel) and bug tracker url does not seem to work, so I decided to post it here. Problem: Errors with mixer controls, breaking alsamixer, and not working 7.1 support. Scenario: Siberia Elite Prism soundcard supports 7.1 channels but it seems it can only play FRONT LEFT and FRONT RIGHT channels, no matter the setup. Here is the alsa-info script output: http://www.alsa-project.org/db/?f=301e6045129665d086d2571652c62648de0eac99 Additionally: aplay -L output: front:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Front speakers surround40:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample mixing device dsnoop:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct sample snooping device hw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Direct hardware device without any conversions plughw:CARD=Elite,DEV=0 SteelSeries Siberia Elite, USB Audio Hardware device with all software conversions and amixer output for this device: # for i in {1..32}; do amixer -cElite cget numid=$i; done numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=2,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | | chmap-fixed=FL,FR numid=3,iface=MIXER,name='Sidetone Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=4,iface=MIXER,name='Sidetone Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=56,step=0 : values=0 | dBminmax-min=-74.00dB,max=-18.00dB numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=10,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB numid=12,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=13,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=74,step=0 : values=73,73 | dBminmax-min=-74.00dB,max=0.00dB amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 amixer: Cannot find the given element from control hw:2 And hw-params from aplay: aplay --dump-hw-params -d1 -Dhw:Elite /dev/zero Playing raw data '/dev/zero' : Unsigned 8 bit, Rate 8000 Hz, Mono HW Params of device "hw:Elite": -------------------- ACCESS: MMAP_INTERLEAVED RW_INTERLEAVED FORMAT: S16_LE S24_3LE SUBFORMAT: STD SAMPLE_BITS: [16 24] FRAME_BITS: [32 192] CHANNELS: [2 8] RATE: [8000 96000] PERIOD_TIME: [1000 16384000] PERIOD_SIZE: [8 131072] PERIOD_BYTES: [64 524288] PERIODS: [2 1024] BUFFER_TIME: (166 32768000] BUFFER_SIZE: [16 262144] BUFFER_BYTES: [64 1048576] TICK_TIME: ALL -------------------- aplay: set_params:1233: Sample format non available Available formats: - S16_LE - S24_3LE Also in syslog I often can see messages related to this device: usb 1-1.3: 10:0: cannot get min/max values for control 3 (id 10) usb 1-1.3: 2:1: cannot get freq at ep 0x1 Loading alsamixer and trying to select the soundcard exits the program with this message: cannot load mixer controls: Invalid argument If anyone wants to check this issue, should you need any type of info regarding this, please do not hesitate asking, I will post anything you require. PS: I don't think it is really necessary to say either but I don't expect anyone to force-commit to this either, I mainly want to know if there is a solution to this or if reporting this issue I can help to get it fixed. Thanks for your time. -- *Daniel Ferradal*

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[alsa-devel] [PATCH] ASoC: Intel: acpi_probe: fix error return path
by Vinod Koul 19 Mar '15

19 Mar '15

Fix the sst_acpi_probe memory allocation error path by setting right error code and initiating the cleanup insteadof just returning Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b782dfdcdbba..b536ddd2517d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -309,7 +309,7 @@ static int sst_acpi_probe(struct platform_device *pdev) ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), GFP_KERNEL); if (!ctx->shim_regs64) { - return -ENOMEM; + ret = -ENOMEM; goto do_sst_cleanup; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/1] Intel: ASoC: Add condition check before set param to waves
by han.lu＠intel.com 19 Mar '15

19 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Check waves state before set parameter through ipc to prevent unexpected operation. Also remove redundant check. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 43fb5f3..20b629a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -2055,6 +2055,11 @@ int sst_hsw_launch_param_buf(struct sst_hsw *hsw) { int ret, idx; + if (!sst_hsw_is_module_active(hsw, SST_HSW_MODULE_WAVES)) { + dev_dbg(hsw->dev, "module waves is not active\n"); + return 0; + } + /* put all param lines to DSP through ipc */ for (idx = 0; idx < hsw->param_idx_w; idx++) { ret = sst_hsw_module_set_param(hsw, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index b40ec74..6c6229a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -399,13 +399,9 @@ static int hsw_waves_param_put(struct snd_kcontrol *kcontrol, if (ret < 0) return ret; - if (sst_hsw_is_module_loaded(hsw, id)) { - if (!sst_hsw_is_module_active(hsw, id)) - return 0; - + if (sst_hsw_is_module_active(hsw, id)) ret = sst_hsw_module_set_param(hsw, id, 0, param_id, param_size, ucontrol->value.bytes.data); - } return ret; } -- 1.9.1

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