[PATCH 1/2] ALSA: add DSD formats

[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@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

[PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams

[Daniel Mack]

Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
typically by any of the new SNDRV_PCM_FMTBIT_DSD_DOP_{U8_U16}, and it
currently does so for USB devices matching the 'Playback Design'
vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
---
 sound/usb/format.c |  3 +++
 sound/usb/quirks.c | 23 +++++++++++++++++++++++
 sound/usb/quirks.h |  4 ++++
 3 files changed, 30 insertions(+)

diff --git a/sound/usb/format.c b/sound/usb/format.c
index a695caf..a73d67a 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -136,6 +136,9 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
 		snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n",
 			   chip->dev->devnum, fp->iface, fp->altsetting, format);
 	}
+
+	pcm_formats |= snd_usb_interface_dsd_format_quirks(chip, fp, sample_bytes);
+
 	return pcm_formats;
 }
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a2ac004..2fcc984 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -905,3 +905,26 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 		mdelay(20);
 }
 
+/*
+ * snd_usb_interface_dsd_format_quirks() is called from format.c to
+ * augment the PCM format bit-field for DSD types. The UAC standards
+ * don't have a designated bit field to denote DSD-capable interfaces,
+ * hence all hardware that is known to support this format has to be
+ * listed here.
+ */
+u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
+					struct audioformat *fp,
+					unsigned int sample_bytes)
+{
+	if ((le16_to_cpu(chip->dev->descriptor.idVendor) == 0x23ba) &&
+	    fp->altsetting > 1) {
+		switch (sample_bytes) {
+		case 1:
+			return SNDRV_PCM_FMTBIT_DSD_DOP_U8;
+		case 2:
+			return SNDRV_PCM_FMTBIT_DSD_DOP_U16;
+		}
+	}
+
+	return 0;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index 0ca9e91..f10d00e 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -30,4 +30,8 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 			   __u8 request, __u8 requesttype, __u16 value,
 			   __u16 index, void *data, __u16 size);
 
+u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
+					struct audioformat *fp,
+					unsigned int sample_bytes);
+
 #endif /* __USBAUDIO_QUIRKS_H */
-- 
1.8.1.4

[PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams

[Daniel Mack]

Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
typically by any of the new SNDRV_PCM_FMTBIT_DSD_DOP_{U8_U16}, and it
currently does so for USB devices matching the 'Playback Design'
vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
---
 sound/usb/format.c |  3 +++
 sound/usb/quirks.c | 23 +++++++++++++++++++++++
 sound/usb/quirks.h |  4 ++++
 3 files changed, 30 insertions(+)

diff --git a/sound/usb/format.c b/sound/usb/format.c
index a695caf..a73d67a 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -136,6 +136,9 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
 		snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n",
 			   chip->dev->devnum, fp->iface, fp->altsetting, format);
 	}
+
+	pcm_formats |= snd_usb_interface_dsd_format_quirks(chip, fp, sample_bytes);
+
 	return pcm_formats;
 }
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a2ac004..2fcc984 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -905,3 +905,26 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 		mdelay(20);
 }
 
+/*
+ * snd_usb_interface_dsd_format_quirks() is called from format.c to
+ * augment the PCM format bit-field for DSD types. The UAC standards
+ * don't have a designated bit field to denote DSD-capable interfaces,
+ * hence all hardware that is known to support this format has to be
+ * listed here.
+ */
+u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
+					struct audioformat *fp,
+					unsigned int sample_bytes)
+{
+	if ((le16_to_cpu(chip->dev->descriptor.idVendor) == 0x23ba) &&
+	    fp->altsetting > 1) {
+		switch (sample_bytes) {
+		case 1:
+			return SNDRV_PCM_FMTBIT_DSD_DOP_U8;
+		case 2:
+			return SNDRV_PCM_FMTBIT_DSD_DOP_U16;
+		}
+	}
+
+	return 0;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index 0ca9e91..f10d00e 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -30,4 +30,8 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
 			   __u8 request, __u8 requesttype, __u16 value,
 			   __u16 index, void *data, __u16 size);
 
+u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
+					struct audioformat *fp,
+					unsigned int sample_bytes);
+
 #endif /* __USBAUDIO_QUIRKS_H */
-- 
1.8.1.4

Re: [PATCH 2/2] ALSA: snd-usb: add quirks handler for DSD streams

[Daniel Mack]

On 06.04.2013 13:44, Daniel Mack wrote:
> Unfortunately, none of the UAC standards provides a way to identify DSD
> (Direct Stream Digital) formats. Hence, this patch adds a quirks
> handler to identify USB interfaces that are capable of handling DSD.
> 
> That quirks handler can augment the already parsed formats bit-field,
> typically by any of the new SNDRV_PCM_FMTBIT_DSD_DOP_{U8_U16}, and it
> currently does so for USB devices matching the 'Playback Design'
> vendor id.

Sorry - this identical message got sent twice unintentionally. Please
ignore one of the two ...


Daniel


> Signed-off-by: Daniel Mack <zonque@gmail.com>
> ---
>  sound/usb/format.c |  3 +++
>  sound/usb/quirks.c | 23 +++++++++++++++++++++++
>  sound/usb/quirks.h |  4 ++++
>  3 files changed, 30 insertions(+)
> 
> diff --git a/sound/usb/format.c b/sound/usb/format.c
> index a695caf..a73d67a 100644
> --- a/sound/usb/format.c
> +++ b/sound/usb/format.c
> @@ -136,6 +136,9 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
>  		snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n",
>  			   chip->dev->devnum, fp->iface, fp->altsetting, format);
>  	}
> +
> +	pcm_formats |= snd_usb_interface_dsd_format_quirks(chip, fp, sample_bytes);
> +
>  	return pcm_formats;
>  }
>  
> diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
> index a2ac004..2fcc984 100644
> --- a/sound/usb/quirks.c
> +++ b/sound/usb/quirks.c
> @@ -905,3 +905,26 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
>  		mdelay(20);
>  }
>  
> +/*
> + * snd_usb_interface_dsd_format_quirks() is called from format.c to
> + * augment the PCM format bit-field for DSD types. The UAC standards
> + * don't have a designated bit field to denote DSD-capable interfaces,
> + * hence all hardware that is known to support this format has to be
> + * listed here.
> + */
> +u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
> +					struct audioformat *fp,
> +					unsigned int sample_bytes)
> +{
> +	if ((le16_to_cpu(chip->dev->descriptor.idVendor) == 0x23ba) &&
> +	    fp->altsetting > 1) {
> +		switch (sample_bytes) {
> +		case 1:
> +			return SNDRV_PCM_FMTBIT_DSD_DOP_U8;
> +		case 2:
> +			return SNDRV_PCM_FMTBIT_DSD_DOP_U16;
> +		}
> +	}
> +
> +	return 0;
> +}
> diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
> index 0ca9e91..f10d00e 100644
> --- a/sound/usb/quirks.h
> +++ b/sound/usb/quirks.h
> @@ -30,4 +30,8 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
>  			   __u8 request, __u8 requesttype, __u16 value,
>  			   __u16 index, void *data, __u16 size);
>  
> +u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
> +					struct audioformat *fp,
> +					unsigned int sample_bytes);
> +
>  #endif /* __USBAUDIO_QUIRKS_H */
> 

Re: [PATCH 1/2] ALSA: add DSD formats

[Jussi Laako]

On 04/06/2013 02:44 PM, Daniel Mack wrote:
> +#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)

No, this is not good, DoP always has the 8-bit marker in the MSB and has 
to be either 24-bit or longer word. One byte marker and two bytes of 
payload, padded with one byte of zeros if payload is 32-bit dword.

Re: [PATCH 1/2] ALSA: add DSD formats

[Daniel Mack]

On 06.04.2013 19:53, Jussi Laako wrote:
> On 04/06/2013 02:44 PM, Daniel Mack wrote:
>> +#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)
> 
> No, this is not good, DoP always has the 8-bit marker in the MSB and has 
> to be either 24-bit or longer word. One byte marker and two bytes of 
> payload, padded with one byte of zeros if payload is 32-bit dword.
> 

Well, the idea was to denote the fact that transport goes in multiple of
8 or 16 bits, which can then be interpreted by the calculation of the
actual sample rate. But I'm fine with anything else. What name would you
pick, assuming that you're fine with the general approach?


Thanks,
Daniel

Re: [PATCH 1/2] ALSA: add DSD formats

[Jussi Laako]

On 04/06/2013 08:59 PM, Daniel Mack wrote:
> On 06.04.2013 19:53, Jussi Laako wrote:
>> On 04/06/2013 02:44 PM, Daniel Mack wrote:
>>> +#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)
>
> Well, the idea was to denote the fact that transport goes in multiple of
> 8 or 16 bits, which can then be interpreted by the calculation of the
> actual sample rate. But I'm fine with anything else. What name would you
> pick, assuming that you're fine with the general approach?

I would just leave out the DOP part and call it SNDRV_PCM_FMTBIT_DSD_U8 
(8 DSD samples per unit) and SDNRV_PCM_FMTBIT_DSD_U16 (16 DSD samples 
per unit). This way it is agnostic to the fact if driver actually 
implements DoP or something else when talking to the actual hardware. It 
assumes all the bits in the unit to be valid DSD samples without any DoP 
marker byte stuff. If driver implements DoP inside, it shouldn't be 
visible outside. If application implements DoP it will anyway pick a 
traditional PCM sample format for the purpose.

It could also  have SNDRV_PCM_FMTBIT_DSD_U32 which is suitable for 
PCI/PCIe and many USB cases. Helps on alignment requirements...

For example SNDRV_PCM_FMTBIT_DSD_U8 and ALSA sample rate of 352800 -> 
DSD sample rate of 2822400. Or SDNRV_PCM_FMTBIT_DSD_U16 and 176400 -> 
DSD sample rate of 2822400.


	- Jussi

Re: [PATCH 1/2] ALSA: add DSD formats

[Jussi Laako]

On 04/06/2013 02:44 PM, Daniel Mack wrote:
>                                                     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

This looks fine, I would just put a comment somewhere like "oldest bit 
is in LSB" or "oldest bit is in MSB". Then for the U16 rest is about 
question of byte endianess for hw vs sw...

Of course, from decoding point of view oldest-in-LSB is easiest since it 
becomes just "y = x & 1; x >>= 1;" loop, while for encoding 
oldest-in-MSB is easiest since it's "x |= y; x <<= 1;"

Re: [PATCH 1/2] ALSA: add DSD formats

[Takashi Iwai]

At Sat,  6 Apr 2013 13:44:51 +0200,
Daniel Mack wrote:
> 
> 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@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) */

What about endianess?  I suppose it's limited only to little-endian?
If so, write it clearly in comments.


> +#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)

Please update pcm_formats[] table in soud/core/pcm_misc.c, too.


thanks,

Takashi

[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 samples
of DSD type through ordinary PCM channels, the "DoP open Standard" was
created. See http://www.dsd-guide.com for a copy of the documentation.

The two new types describe streams that are capable of handling DSD
samples in 8-bit or in 16-bit (or at a single or double data rate,
respectively). For applications, the following mapping table is used
to configure the DSD mode by selecting a combination of PCM sample
rates and word lengths:

        352.8kHz       705.6KHz     1411.2KHz

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

Signed-off-by: Daniel Mack <zonque@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..2dffd55 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_U8		_SNDRV_PCM_FMTBIT(DSD_U8)
+#define SNDRV_PCM_FMTBIT_DSD_U16	_SNDRV_PCM_FMTBIT(DSD_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..f9b0d3f 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_U8		((__force snd_pcm_format_t) 48) /* DSD in 1 byte */
+#define	SNDRV_PCM_FORMAT_DSD_U16	((__force snd_pcm_format_t) 49) /* DSD in 2 bytes */
+#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_DSD_16U
 
 #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..433efdb 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_U8),
+	FORMAT(DSD_U16),
 };
 
 const char *snd_pcm_format_name(snd_pcm_format_t format)
-- 
1.8.1.4