[alsa-devel] [PATCH 1/2] ALSA: add DSD formats

[Daniel Mack]

This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital

DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream. In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for a copy of the documentation.

The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).

DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:

                                                   configured hardware
        352.8kHz       705.6KHz     1411.2KHz  <----       sample rate

8-bit     2.8MHz         5.6MHz       11.2MHz
16-bit    5.6MHz        11.2MHz

         `-----------------------------------'
                 actual DSD sample rates

Signed-off-by: Daniel Mack <zonque at gmail.com>
---
 include/sound/pcm.h         | 2 ++
 include/uapi/sound/asound.h | 4 +++-
 sound/core/pcm.c            | 2 ++
 3 files changed, 7 insertions(+), 1 deletion(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index aa7b0a8..d957046 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -181,6 +181,8 @@ struct snd_pcm_ops {
 #define SNDRV_PCM_FMTBIT_G723_24_1B	_SNDRV_PCM_FMTBIT(G723_24_1B)
 #define SNDRV_PCM_FMTBIT_G723_40	_SNDRV_PCM_FMTBIT(G723_40)
 #define SNDRV_PCM_FMTBIT_G723_40_1B	_SNDRV_PCM_FMTBIT(G723_40_1B)
+#define SNDRV_PCM_FMTBIT_DSD_DOP_U8	_SNDRV_PCM_FMTBIT(DSD_DOP_U8)
+#define SNDRV_PCM_FMTBIT_DSD_DOP_U16	_SNDRV_PCM_FMTBIT(DSD_DOP_U16)
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define SNDRV_PCM_FMTBIT_S16		SNDRV_PCM_FMTBIT_S16_LE
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 1774a5c..eb9eda8 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t;
 #define	SNDRV_PCM_FORMAT_G723_24_1B	((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */
 #define	SNDRV_PCM_FORMAT_G723_40	((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */
 #define	SNDRV_PCM_FORMAT_G723_40_1B	((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */
-#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_G723_40_1B
+#define	SNDRV_PCM_FORMAT_DSD_DOP_U8	((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD DOP format (x8) */
+#define	SNDRV_PCM_FORMAT_DSD_DOP_U16	((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD DOP format (x16) */
+#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_DSD_DOP_U16
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define	SNDRV_PCM_FORMAT_S16		SNDRV_PCM_FORMAT_S16_LE
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 578327e..578a761 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = {
 	FORMAT(G723_24_1B),
 	FORMAT(G723_40),
 	FORMAT(G723_40_1B),
+	FORMAT(DSD_DOP_U8),
+	FORMAT(DSD_DOP_U16),
 };
 
 const char *snd_pcm_format_name(snd_pcm_format_t format)
-- 
1.8.1.4