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? 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 | _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
