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

[alsa-devel] Thinkpad T400 - internal mic doesn't work, can't set CAPTURE - schematics included
by Arkadiusz Miśkiewicz 01 Mar '13

01 Mar '13

Hi, My Thinkpad T400 (2764CTO model) doesn't have ability to set CAPTURE on microphone inputs with alsa. Most likely some quirk is missing. The laptop has internal microphone, microphone in input, headphones out output and 2 internal speakers. That's all. Is uses Conexant CX20561_15Z internally and there is detailed T400 schematic available: http://kythuatphancung.com/download/lenovo-thinkpad-t400-schematic- diagram.html?dl (for audio look around page 43). Kernels 3.7.x, fresh 3.8 - no luck. Tried even linux-next from today but it dies early on alsa unrelated issues. I've tried model options laptop, hp, hp-dv6736, hp-700, toshiba, ideapad, auto. None of these work fine. With model=hp-700 I can set CAPTURE but it doesn't work (silent audio is captured using arecord or in google hangout). External mic connected also doesn't work with these setups. http://www.alsa-project.org/db/?f=394ff4cdca33e75c8a47203d42b89d8de36deca7 [3.8 kernel] http://www.alsa-project.org/db/?f=302877e50f99835075d038207f6db0dfd89873b0 [3.7 kernel] Thinkpad is 2764CTO model exactly. I can test various hacks, patches etc easily. Could someone look at schematic and provide quirks to test? Thanks! -- Arkadiusz Miśkiewicz, arekm / maven.pl

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[alsa-devel] [PATCH v3] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 01 Mar '13

01 Mar '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. For TDM mode, McASP is capable to play or receive 32 channels for one serializer. McAsp has support for max 16 serializer, therefore max channels is 32 * 8. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 66 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 +++++---- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 68 insertions(+), 15 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..0a1af1a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ 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 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ 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 < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + 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 < max_active_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 < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + 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); + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ 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 + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, @@ -1015,8 +1057,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->op_mode = val; ret = of_property_read_u32(np, "tdm-slots", &val); - if (ret >= 0) + if (ret >= 0) { + if (val < 2 || val > 32) { + dev_err(&pdev->dev, + "tdm-slots must be in rage [2-32]\n"); + ret = -EINVAL; + goto nodata; + } + pdata->tdm_slots = val; + } ret = of_property_read_u32(np, "num-serializer", &val); if (ret >= 0) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 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; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { 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 +225,10 @@ 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, + fifo_level * 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 1/2] ASoC: wm5102: Correct OUT2 volume and switch names
by Charles Keepax 01 Mar '13

01 Mar '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83..deb1097 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -755,7 +755,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +767,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: arizona: Don't enable FLL on REFCLK configuration
by Charles Keepax 01 Mar '13

01 Mar '13

Enabling the FLL when REFCLK is being configured is not what the user would expect and can cause issues if SYNCCLK has no specified frequency. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 15 ++++++--------- 1 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6837863..eb25ab9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1148,17 +1148,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (source < 0) return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) + if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (fll->fout) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); if (ret != 0) return ret; if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, Fout); + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); if (ret != 0) return ret; } @@ -1166,12 +1166,9 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - fll->fout = Fout; - if (Fout) { + if (fll->fout) { arizona_enable_fll(fll, &ref, &sync); - } else { - arizona_disable_fll(fll); } return 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] hdajackretask: Fix "advanced override" bug
by David Henningsson 01 Mar '13

01 Mar '13

The values were not correctly retreived from the comboboxes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 65d0f3a..4227669 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -183,14 +183,14 @@ static void update_user_pin_config(ui_data_t* ui) continue; if (ui->free_overrides) { - int i; + int j; int index; unsigned long val = 0; - for (i = 0; i < FREE_OVERRIDES_COUNT; i++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); + if (index < 0) break; - val += get_free_override_list(i)[index].value; + val += get_free_override_list(j)[index].value; } if (index < 0) continue; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: snd-core: tidyup snd_soc_register_codec() fail case
by Kuninori Morimoto 01 Mar '13

01 Mar '13

kfree() on snd_soc_register_codec() was summarized to one place. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 689eb04..ceac893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3983,8 +3983,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4023,7 +4023,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4057,8 +4057,9 @@ int snd_soc_register_codec(struct device *dev, dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: kfree(codec->name); +fail_codec: kfree(codec); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Use LPIB position fix for Acer AO751h
by David Henningsson 28 Feb '13

28 Feb '13

The user reports a regression (choppy audio), and that position_fix=1 (or position_fix=4) fixes the issue. This is a Poulsbo controller (8086:811b (rev 07)), so might be worth checking if other Poulsbo machines suffer from the same bug. Cc: stable(a)vger.kernel.org [v3.7+] BugLink: https://bugs.launchpad.net/bugs/1132562 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6..3505f1e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3114,6 +3114,7 @@ static bool check_hdmi_disabled(struct pci_dev *pci) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] = { + SND_PCI_QUIRK(0x1025, 0x0244, "Acer AO751h", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), -- 1.7.9.5

2 5

[alsa-devel] Query about SNDRV_PCM_FORMAT_S24_LE
by Patrick Lai 28 Feb '13

28 Feb '13

Hi, The DSP I am working with currently supports 24-bit PCM sample in 32-bit data space but with LSB padded. Based on the discussion I found on line, SNDRV_PCM_FORMAT_S24_LE is defined to be 24-bit PCM sample in 32-bit data space with MSB padded. Also, there is no format macro defined for the format which my DSP supports. Please correct me if I am wrong. Furthermore, which software layer in audio stack of a typical linux distribution such as ubuntu handles the padding. I don't think it's handled by the alsa library. Does application or pulseaudio handle the padding? I would think it's done in pulseaudio. I have not study pulseaudio codec extensively but would think someone on the mailing list can give a quick answer. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[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 | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..9f33d90 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ 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 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ 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 < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + 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 < max_active_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 < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + 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); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ 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 + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ 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..3af8b50 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; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { 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 +225,10 @@ 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, + fifo_level * 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 10/12] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 27 Feb '13

27 Feb '13

With the generic DMA device tree helper supported by mxs-dma driver, client devices only need to call dma_request_slave_channel() for requesting a DMA channel from dmaengine. Call snd_dmaengine_pcm_open() with NULL filter_fn, and having parameters needed by dma_request_slave_channel() wrap into snd_dma_channel_params and pass it in as filter_data, so that dmaengine_pcm will call the helper rather than dma_request_channel() to request DMA channel. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- .../devicetree/bindings/sound/mxs-saif.txt | 18 ++++---- sound/soc/mxs/mxs-pcm.c | 43 ++------------------ sound/soc/mxs/mxs-pcm.h | 5 --- sound/soc/mxs/mxs-saif.c | 28 ++----------- sound/soc/mxs/mxs-saif.h | 3 +- 5 files changed, 20 insertions(+), 77 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..da221b4 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "saif". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,15 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "saif"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; - fsl,saif-master = <&saif0>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "saif"; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 564b5b6..74cb4b5 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -28,7 +28,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dmaengine.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/initval.h> @@ -39,11 +38,6 @@ #include "mxs-pcm.h" -struct mxs_pcm_dma_data { - struct mxs_dma_data dma_data; - struct mxs_pcm_dma_params *dma_params; -}; - static struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -64,22 +58,6 @@ static struct snd_pcm_hardware snd_mxs_hardware = { }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_data *pcm_dma_data = param; - struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &pcm_dma_data->dma_data; - - return true; -} - static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -91,35 +69,22 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_mxs_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mxs_pcm_dma_data *pcm_dma_data; + struct snd_dma_channel_params *params; int ret; - pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); - if (pcm_dma_data == NULL) - return -ENOMEM; - - pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq; - - ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data); - if (ret) { - kfree(pcm_dma_data); + params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + ret = snd_dmaengine_pcm_open(substream, NULL, params); + if (ret) return ret; - } snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); - snd_dmaengine_pcm_set_data(substream, pcm_dma_data); - return 0; } static int snd_mxs_close(struct snd_pcm_substream *substream) { - struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); - snd_dmaengine_pcm_close(substream); - kfree(pcm_dma_data); return 0; } diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 35ba2ca..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,11 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -struct mxs_pcm_dma_params { - int chan_irq; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3a2aa1d..9f6abb8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -659,7 +658,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; struct pinctrl *pinctrl; int ret = 0; @@ -721,22 +720,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -753,13 +736,8 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.chan_irq < 0) { - ret = saif->dma_param.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } + saif->dma_param.dev = saif->dev; + saif->dma_param.name = "saif"; platform_set_drvdata(pdev, saif); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..32b4310 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -108,6 +108,7 @@ #define MXS_SAIF_MCLK 0 +#include <sound/dmaengine_pcm.h> #include "mxs-pcm.h" struct mxs_saif { @@ -117,7 +118,7 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; + struct snd_dma_channel_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

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