[alsa-devel] Device start to Play Native DSD in SoC framwork (WAS Re: DSD native format on SoC)
Michael Trimarchi
michael at amarulasolutions.com
Sun Apr 9 21:06:09 CEST 2017
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 at amarulasolutions.com> 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 at amarulasolutions.com
> >>>Cc: alsa-devel at alsa-project.org
> >>>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 at melcoinc.co.jp
> >>>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 |
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