February 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 27 Feb '13

27 Feb '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++------ sound/soc/davinci/davinci-pcm.c | 15 ++++++------ sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..ebd6ee0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -657,12 +657,14 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 stream_serializers = channels/2 + channels%2; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +682,44 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + ser = tx_ser; + } else { + ser = rx_ser; + } + + if (ser < stream_serializers) { + printk(KERN_WARNING "davinci-mcasp: stream has more channels " + "(%d) than are enabled in mcasp (%d)", + channels, ser * 2); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < stream_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < stream_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + 3); } } @@ -812,8 +842,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + davinci_hw_common_param(dev, substream->stream, channels); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +896,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = channels/2 + channels%2; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +971,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..9951973 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -193,7 +194,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) period_size); data_type = prtd->params->data_type; - count = period_size / data_type; + count = period_size / (data_type * serializers); if (fifo_level) count /= fifo_level; @@ -201,8 +202,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,9 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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[alsa-devel] [PATCH v4 0/3] ASoC davinci/tlv320aic3x updates, fixes & DT support
by Hebbar Gururaja 27 Feb '13

27 Feb '13

1. Convert tlv320aic3x mic bias to a supply widget and related machine driver 2. Add DT support for Davinci machine platform 3. Remove __dev* attributes This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130128 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://github.com/ghebbar/linux.git Branch : dev_next-20130128_davinci_asoc_dt_v3 Changes in v4 - updated pending machine driver to change in micbias -> supply widget - merge codec & machine driver change into a single commit Changes in v3 - drop mcasp pinctrl handler patch from series as it is now handled centrally. - Convert tlv320aic3x mic bias to a supply widget and related machine driver - remove __dev* attributes Changes in v2 - Remove reference to Linux & software details from DT binding Hebbar Gururaja (2): ASoC: tlv320aic3x: Convert mic bias to a supply widget ASoC: davinci: remove __dev* attributes Hebbar, Gururaja (1): ASoC: davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + include/sound/tlv320aic3x.h | 10 ++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++--- sound/soc/omap/n810.c | 4 +- sound/soc/omap/rx51.c | 8 +- 8 files changed, 318 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct AEC loopback enable definiton
by Mark Brown 26 Feb '13

26 Feb '13

Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c1ea75c..efd7317 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1083,7 +1083,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, -- 1.7.10.4

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[alsa-devel] [PATCH - alsa-lib/mixer 1/1] snd_mixer_poll_descriptors_revents should loop over pollfd
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> Signed-off-by: Sander Jansen <s.jansen(a)gmail.com> diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..8deceac 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -743,7 +743,7 @@ int snd_mixer_poll_descriptors_revents(snd_mixer_t *mixer, struct pollfd *pfds, if (nfds == 0) return -EINVAL; res = 0; - for (idx = 0; idx < nfds; idx++) + for (idx = 0; idx < nfds; idx++, pfds++) res |= pfds->revents & (POLLIN|POLLERR|POLLNVAL); *revents = res; return 0; -- 1.8.1.3

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[alsa-devel] [PATCH] Fix: pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> --- pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 4226d30..0d3704b 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -651,6 +651,8 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, else *revents = 0; + err = 0; + finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); -- 1.8.1.4

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[alsa-devel] [PATCH] pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> According to the DOCS, poll_revents should return 0 on success. Fix the pulse plugin to do so. Sander Jansen (1): Fix: pulse_pcm_poll_revents should return 0 on success pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: dapm: Provide early event callbacks for power up and down
by Mark Brown 26 Feb '13

26 Feb '13

Some devices may benefit from being able to start some parts of the widget power up/down sequence earlier on in the sequence than the point at which the final power state is committed. Support these by providing events which are called before any power state changes are done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 38fd652..7e4efa5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -311,6 +311,8 @@ struct device; #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ +#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ +#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ #define SND_SOC_DAPM_PRE_POST_PMD \ (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 11047ba..b69e9de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1255,6 +1255,14 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, ev_name = "POST_PMD"; power = 0; break; + case SND_SOC_DAPM_WILL_PMU: + ev_name = "WILL_PMU"; + power = 1; + break; + case SND_SOC_DAPM_WILL_PMD: + ev_name = "WILL_PMD"; + power = 0; + break; default: BUG(); return; @@ -1715,6 +1723,14 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); + list_for_each_entry(w, &down_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); + } + + list_for_each_entry(w, &up_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + } + /* Power down widgets first; try to avoid amplifying pops. */ dapm_seq_run(dapm, &down_list, event, false); -- 1.7.10.4

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[alsa-devel] [PATCH] Allow to switch hardware sampe rate on EMU
by Mihail Zenkov 25 Feb '13

25 Feb '13

Commit 57e5c63007955838043e34c732d224b2cbbb128f "emu10k1: allow to disable the SRC" force hardware use only one rate (48000 hz), but EMU have support two hardware sampling rates (44100 and 48000 hz). Signed-off-by: Mihail Zenkov <mihail.zenkov(a)gmail.com> --- linux-3.8/sound/pci/emu10k1/emupcm.c +++ linux-3.8-1/sound/pci/emu10k1/emupcm.c @@ -1127,7 +1127,7 @@ struct snd_emu10k1_pcm *epcm; struct snd_emu10k1_pcm_mixer *mix; struct snd_pcm_runtime *runtime = substream->runtime; - int i, err; + int i, err, sample_rate; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) @@ -1146,7 +1146,11 @@ kfree(epcm); return err; } - err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (emu->card_capabilities->emu_model && emu->emu1010.internal_clock == 0) + sample_rate = 44100; + else + sample_rate = 48000; + err = snd_pcm_hw_rule_noresample(runtime, sample_rate); if (err < 0) { kfree(epcm); return err;

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[alsa-devel] M-Audio Audiophile 192 (ice1724)'s broken spdif capture
by Jonas Petersen 25 Feb '13

25 Feb '13

Hi there, I'm trying to get spdif capture to work properly on the M-Audio Audiophile 192 (VT1724/Envy24HT). It is generally working, but I don't get a clean signal. I have "Multi Track Internal Clock" set to "IEC958 In" (spdif in). When I capture via Jack, arecord or Audacity I experience the following two phenomenons (always): 1.) The captured signal ends up 6 dB(FS) to loud (200%)! Everything above -6 dB is distorting. A 1 kHz sine test signal at -12 dB (25%) will be captured as -6 dB (50%). 2.) The left and right channels are shifted by one sample! When I feed a 1 kHz test signal (L+R are _exactly_ the same signal), the right channel will be offset by exactly one sample. Zooming into the waveform clearly shows that. Analyzing the signal with a goniometer shows a (slight but obvious) vertical eliptical shape and not the expected single vertical line. To make sure the hardware actually works fine, I did install a Windows 7 on the exact same machine, installed the drivers from M-Audio and did some recording with audacity. The result is as expected: the -12 dB signal ends up as -12 dB and the left and right channel exactly match each other. So the hardware is willing. I am running a Lubuntu 12.10 and I'm am able to compile and run the current alsa-driver source with the 3.5.0-22 kernel. I played around a bit with the alsa driver source (e.g. pci/ice1712/ice1724.c, pci/ice1712/revo.c) and I am able to compile and load a modified driver. So far I only was able to make the problem worse though. I'd now like to ask for some advice on how to approach the problem. I guess the fact that the left and right channel differ - even though they shouldn't - might be a thing to look for. This must be happening at some stage in the capturing. Is there a way to hook in at different places to narrow down what causes this? Maybe this even already rings somebody's bells? I'll be glad to deliver more information when needed. Thanks and regards, Jonas

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[alsa-devel] [PATCH 1/1] M-Audio Audiophile192: Fix SPDIF input
by Jonas Petersen 25 Feb '13

25 Feb '13

This patch fixes initialization of the AK4114 chip so spdif capture is working properly. Worked out together with Pavel Hofman. Signed-off-by: Jonas Petersen <jnsptrsn1(a)gmail.com> diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 7641080..1112ec1 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -35,6 +35,7 @@ struct revo51_spec { struct snd_i2c_device *dev; struct snd_pt2258 *pt2258; + struct ak4114 *ak4114; }; static void revo_i2s_mclk_changed(struct snd_ice1712 *ice) @@ -359,9 +360,9 @@ static struct snd_ak4xxx_private akm_ap192_priv = { .cif = 0, .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, - .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1, - .cs_addr = VT1724_REVO_CS1, - .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3, + .cs_addr = VT1724_REVO_CS3, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, }; @@ -372,7 +373,7 @@ static struct snd_ak4xxx_private akm_ap192_priv = { * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358) * CSN (pin 35) -- GPIO7 pin 59 */ -#define AK4114_ADDR 0x02 +#define AK4114_ADDR 0x00 static void write_data(struct snd_ice1712 *ice, unsigned int gpio, unsigned int data, int idx) @@ -426,7 +427,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) tmp = snd_ice1712_gpio_read(ice); tmp |= VT1724_REVO_CCLK; /* high at init */ tmp |= VT1724_REVO_CS0; - tmp &= ~VT1724_REVO_CS1; + tmp &= ~VT1724_REVO_CS3; snd_ice1712_gpio_write(ice, tmp); udelay(1); return tmp; @@ -434,7 +435,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp) { - tmp |= VT1724_REVO_CS1; + tmp |= VT1724_REVO_CS3; tmp |= VT1724_REVO_CS0; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -470,27 +471,32 @@ static unsigned char ap192_ak4114_read(void *private_data, unsigned char addr) static int ap192_ak4114_init(struct snd_ice1712 *ice) { static const unsigned char ak4114_init_vals[] = { - AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, + AK4114_RST | AK4114_PWN | AK4114_OCKS0, AK4114_DIF_I24I2S, AK4114_TX1E, - AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(1), + AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(0), 0, 0 }; static const unsigned char ak4114_init_txcsb[] = { 0x41, 0x02, 0x2c, 0x00, 0x00 }; - struct ak4114 *ak; int err; + struct revo51_spec *spec; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + err = snd_ak4114_create(ice->card, ap192_ak4114_read, ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, - ice, &ak); + ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - ak->check_flags = AK4114_CHECK_NO_RATE; + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; return 0; /* error ignored; it's no fatal error */ } @@ -562,6 +568,9 @@ static int revo_init(struct snd_ice1712 *ice) ice); if (err < 0) return err; + err = ap192_ak4114_init(ice); + if (err < 0) + return err; /* unmute all codecs */ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, @@ -575,7 +584,7 @@ static int revo_init(struct snd_ice1712 *ice) static int revo_add_controls(struct snd_ice1712 *ice) { - struct revo51_spec *spec; + struct revo51_spec *spec = ice->spec; int err; switch (ice->eeprom.subvendor) { @@ -597,7 +606,9 @@ static int revo_add_controls(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; - err = ap192_ak4114_init(ice); + /* only capture SPDIF over AK4114 */ + err = snd_ak4114_build(spec->ak4114, NULL, + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); if (err < 0) return err; break; -- 1.7.10.4

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