[alsa-devel] DSD native format on SoC
Naoki Matsumoto
n-matsumoto at melcoinc.co.jp
Wed Mar 22 03:43:24 CET 2017
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
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 |
>
>
>
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