Hi all On Wed, Mar 22, 2017 at 11:43:24AM +0900, Naoki Matsumoto wrote: > Hello > > On 2017/03/21 20:29, Michael Nazzareno Trimarchi wrote: > >Hi > > > > > >On Sat, Mar 18, 2017 at 12:14 PM, Michael Trimarchi > ><michael@xxxxxxxxxxxxxxxxxxxx> wrote: > >>Hi all > >> > >>On Mon, Mar 13, 2017 at 05:42:43PM +0900, Naoki Matsumoto wrote: > >>>Retry send because alsa-dev ML doesn't delivery yet. > >>> > >>>-------- Original Message -------- > >>>From: Naoki Matsumoto > >>>Sent: Monday, Mar 13, 2017 10:39 AM GMT+0900 > >>>To: michael@xxxxxxxxxxxxxxxxxxxx > >>>Cc: alsa-devel@xxxxxxxxxxxxxxxx > >>>Subject: DSD native format on SoC > >>> > >>>Hello > >>> > >>>I know only xmos-native-dsd(DSD_U32_BE). > >>>I'll share what I know. > >>>I know information about DSD only fragmentally... > >>> > >> > >>Problem here is SoC subsytem for me > >> > >>https://github.com/zonque/alsa-dsd-player.git > >> > >>Cannot set sample format tyring U16_LE DSD. I have changed a bit the core > >>and some components but seems that does not go in. > >> > >>Any idea? > > > > > >After dig a bit more with strace and fix a typo here, look like that > >available formats are not pass up to the alsa lib. So alsa lib fail > >without even try to go there when it sets format. Can someone suggest > >how to debug in a good way this scenario? > > > >Michael > > Could you check this function? > This function is show up available formats list. > Could you create small program that support format checker? > > show_available_sample_formats(snd_pcm_hw_params_t* params)@aplay.c > http://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=aplay/aplay.c;h=ee480f29b760fb65fd6c5670d79899538b6497d6;hb=1314abd2d61877a92e5289452dee308e98dab0c1#l1178 > > and one more. > if you have decoded sound file, you can try playback using aplay(1) > e,g) aplay -D "hw:0,0" -f DSD_U16_LE -r $(rate_num) -c2 dsd.raw > I think that we have the device playing DSD over I2S. I will post playing music session soon using mpd ;) Michael > Naoki > By the way. > I feel good so you received this mail. > My posts doesn't appear at alsa-lib ML. > I am sad and worried. > > > >> > >>diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c > >>index cc34b16..0dea529 100644 > >>--- a/sound/soc/codecs/pcm179x.c > >>+++ b/sound/soc/codecs/pcm179x.c > >>@@ -89,18 +89,19 @@ static int pcm179x_startup(struct snd_pcm_substream *substream, > >> { > >> struct snd_soc_codec *codec = dai->codec; > >> struct pcm179x_private *priv = snd_soc_codec_get_drvdata(codec); > >>- u64 formats = PCM1792A_FORMATS; > >>+ u64 formats = PCM1795_FORMATS; > >> > >> switch (priv->codec_model) { > >>- case PCM1795: > >>- formats = PCM1795_FORMATS; > >>+ case PCM1792A: > >>+ formats = PCM1792A_FORMATS; > >> break; > >> default: > >> break; > >> } > >> > >>- snd_pcm_hw_constraint_mask64(substream->runtime, > >>- SNDRV_PCM_HW_PARAM_FORMAT, formats); > >>+ if (formats != PCM1795_FORMATS) > >>+ snd_pcm_hw_constraint_mask64(substream->runtime, > >>+ SNDRV_PCM_HW_PARAM_FORMAT, formats); > >> > >> msleep(50); > >> return 0; > >>@@ -227,7 +228,7 @@ static struct snd_soc_dai_driver pcm179x_dai = { > >> .rates = SNDRV_PCM_RATE_CONTINUOUS, > >> .rate_min = 10000, > >> .rate_max = 200000, > >>- .formats = PCM179X_FORMATS, }, > >>+ .formats = PCM1795_FORMATS, }, > >> .ops = &pcm179x_dai_ops, > >> }; > >> > >>@@ -252,9 +253,9 @@ static struct snd_soc_codec_driver soc_codec_dev_pcm179x = { > >> }; > >> > >> const struct of_device_id pcm179x_of_match[] = { > >>- { .compatible = "ti,pcm1792a", }, > >>- { .compatible = "ti,pcm1795", .data = (void *)PCM1795, }, > >>- { .compatible = "ti,pcm1796", }, > >>+ { .compatible = "ti,pcm1792a", .data = (void *)PCM1792A }, > >>+ { .compatible = "ti,pcm1795", }, > >>+ { .compatible = "ti,pcm1796", .data = (void *)PCM1792A }, > >> { } > >> }; > >> MODULE_DEVICE_TABLE(of, pcm179x_of_match); > >>diff --git a/sound/soc/codecs/pcm179x.h b/sound/soc/codecs/pcm179x.h > >>index 4c00047..0665ec8 100644 > >>--- a/sound/soc/codecs/pcm179x.h > >>+++ b/sound/soc/codecs/pcm179x.h > >>@@ -22,7 +22,8 @@ > >> > >> #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) > >> > >>-#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE) > >>+#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ > >>+ SNDRV_PCM_FMTBIT_DSD_U16_LE) > >> > >> extern const struct regmap_config pcm179x_regmap_config; > >> extern const struct of_device_id pcm179x_of_match[]; > >>diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig > >>index c55cbfa..d3afa89 100644 > >>--- a/sound/soc/rockchip/Kconfig > >>+++ b/sound/soc/rockchip/Kconfig > >>@@ -55,7 +55,7 @@ config SND_SOC_ROCHCHIP_DACMAX > >> tristate "SoC Audio support for DACMAX boards using a pcm1792 codec" > >> depends on SND_SOC_ROCKCHIP && SPI && GPIOLIB > >> select SND_SOC_ROCKCHIP_I2S > >>- select SND_SOC_PCM179X > >>+ select SND_SOC_PCM179X_SPI > >> help > >> Say Y or M here if you want to add support for SoC audio on Dacmax > >> boards using the pcm1792a codec. > >>diff --git a/sound/soc/rockchip/dacmax.c b/sound/soc/rockchip/dacmax.c > >>index fb302a8..fa79549 100644 > >>--- a/sound/soc/rockchip/dacmax.c > >>+++ b/sound/soc/rockchip/dacmax.c > >>@@ -22,6 +22,7 @@ > >> * > >> */ > >> > >>+#define DEBUG > >> #include <linux/module.h> > >> #include <linux/of.h> > >> #include <linux/of_gpio.h> > >>@@ -39,6 +40,7 @@ > >> #define CLK1 (1 << 1) > >> #define CLK0 (1 << 2) > >> #define W32 (1 << 3) > >>+#define DSD_EN (1 << 4) > >> > >> #define DAI_NAME_SIZE 32 > >> > >>@@ -52,6 +54,7 @@ struct dacmax_data { > >> int clk_1; > >> int clk_2; > >> int w32; > >>+ int dsd_enable; > >> }; > >> > >> static const struct snd_soc_dapm_widget dacmax_dapm_widgets[] = { > >>@@ -78,6 +81,10 @@ static void dacmax_change_freq(struct dacmax_data *data, u8 mask) > >> gpio_set_value(data->w32, value); > >> pr_debug("%s: BITSXWORD(%d)\n", __func__, value); > >> > >>+ value = (mask & DSD_EN) ? 1 : 0; > >>+ gpio_set_value(data->dsd_enable, value); > >>+ pr_debug("%s: DSD ENABLE (%d)\n", __func__, value); > >>+ > >> mdelay(20); > >> } > >> > >>@@ -91,6 +98,7 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, > >> params_format(params)); > >> > >> switch (params_format(params)) { > >>+ case SNDRV_PCM_FORMAT_DSD_U16_LE: > >> case SNDRV_PCM_FORMAT_S16_LE: > >> break; > >> case SNDRV_PCM_FORMAT_S24_LE: > >>@@ -103,6 +111,8 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, > >> } > >> > >> switch (params_rate(params)) { > >>+ case 2822400: > >>+ mask |= DSD_EN; > >> case 44100: > >> break; > >> case 48000: > >>@@ -251,6 +261,7 @@ static int dacmax_probe(struct platform_device *pdev) > >> data->clk_1 = clk_1; > >> data->clk_2 = clk_2; > >> data->w32 = w32; > >>+ data->dsd_enable = dsd_enable; > >> > >> ret = devm_snd_soc_register_card(&pdev->dev, &data->card); > >> if (ret) { > >>diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c > >>index 68fea0a..67fd6ec 100644 > >>--- a/sound/soc/rockchip/rockchip_i2s.c > >>+++ b/sound/soc/rockchip/rockchip_i2s.c > >>@@ -302,6 +302,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, > >> case SNDRV_PCM_FORMAT_S8: > >> val |= I2S_TXCR_VDW(8); > >> break; > >>+ case SNDRV_PCM_FORMAT_DSD_U16_LE: > >> case SNDRV_PCM_FORMAT_S16_LE: > >> val |= I2S_TXCR_VDW(16); > >> break; > >>@@ -457,7 +458,8 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { > >> SNDRV_PCM_FMTBIT_S16_LE | > >> SNDRV_PCM_FMTBIT_S20_3LE | > >> SNDRV_PCM_FMTBIT_S24_LE | > >>- SNDRV_PCM_FMTBIT_S32_LE), > >>+ SNDRV_PCM_FMTBIT_S32_LE | > >>+ SNDRV_PCM_FMTBIT_DSD_U16_LE), > >> }, > >> .capture = { > >> .stream_name = "Capture", > >>diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > >>index a1305f8..1ae0aea 100644 > >>--- a/sound/soc/soc-core.c > >>+++ b/sound/soc/soc-core.c > >>@@ -2980,6 +2980,7 @@ static u64 codec_format_map[] = { > >> SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE, > >> SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE, > >> SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE, > >>+ SNDRV_PCM_FORMAT_DSD_U16_LE | SNDRV_PCM_FORMAT_DSD_U16_BE, > >> SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE > >> | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, > >> }; > >>>on Wed, 8 Mar 2017 08:21:23 +0100, Michael Nazzareno Trimarchi wrote: > >>>DSD is a continuous stream of bits that flows over two separate > >>>channels, left and right, synchronized by a clock, while I2S is a > >>>single data wire and an additional wire that states if the incoming > >>>sample refers to the left channel or the right channel. There is no > >>>way to get DSD data other than a circuit which decode DSD streams > >>>coming directly from a conventional source, like, for examples, a > >>>SACD. > >>> > >>>The Volta dac has an internal decoding circuitry that still employs > >>>the I2S standard as a usual 44100Hz/16 bits, but sourcing the shift > >>>clock at twice the DSD standard frequency of 2.8224MHz, since it has > >>>to split two the16 bits words received on a single wire into the two > >>>DSD channels, as the standard wants. > >>> > >>>When DSD is enabled, the control signal PCD/DSD must be high, CK3..CK0 > >>>must be all low. The I2S interface works as a standard 44100Hz/16 bit > >>>and the DSD streaming must be packed into 16 bits lenght words > >>>left/right as per LRCK logic. The BCLK frequency supplied from the > >>>interface is 5644800Hz in case of DSD and 11289600 for DSD2. > >>>I don't know your device. > >>>but I've understood that it's device layer implantation topic. > >>>I think that your sound device need to support SND_PCM_FORMAT_DSD_*. > >>> > >>>Refer:enum snd_pcm_format_t > >>>http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > >>> > >>>If you use xmos-natvie-dsd, the implement is linux.git/sound/usb/quirks.c > >>> > >>>Can you explain how now DSD can be pass to audio card? The idea is to > >>>declare it on the audio card and then configure it as a PCM card but I > >>>don't know how data are suppose to arrive from alsa userspace. > >>>I know DSD playback on sound device that xmos. > >>>Have you checked this page? > >>>https://github.com/lintweaker/xmos-native-dsd > >>> > >>>As alsa-lib > >>>1. snd_pcm_open > >>>e.g, USB sound device. In your case, soc internal sound device > >>>2. snd_ocm_hw_params_set_format > >>>e.g, SND_PCM_FORMAT_DSD_U32_BE > >>>3. snd_pcm_writei > >>> > >>>Other elements > >>>* DSD decoder(e.g,dsf/dsdiff) > >>>* Player > >>>if you use xmos-native-dsd(DSD_U32_BE) device, we can use MPD Ver0.20.2+ > >>>I don't know other DSD pcm_formats. > >>> > >>>Just information. It may be wrong. > >>>Thank you > >>> > >>> > >>>-- > >>>********************************************** > >>>Naoki MATSUMOTO > >>>Email:n-matsumoto@xxxxxxxxxxxxxx > >>>Tel :050-5830-8916 > >>>********************************************** > >> > >>-- > >>| Michael Nazzareno Trimarchi Amarula Solutions BV | > >>| COO - Founder Cruquiuskade 47 | > >>| +31(0)851119172 Amsterdam 1018 AM NL | > >>| [`as] http://www.amarulasolutions.com | > > > > > > > -- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL | | [`as] http://www.amarulasolutions.com | _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
