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