From mboxrd@z Thu Jan 1 00:00:00 1970 From: Daniel Mack Subject: [PATCH 1/2] ALSA: add DSD formats Date: Thu, 28 Mar 2013 00:30:45 +0100 Message-ID: <1364427046-29944-1-git-send-email-zonque@gmail.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail.zonque.de (svenfoo.org [82.94.215.22]) by alsa0.perex.cz (Postfix) with ESMTP id 884FA26531B for ; Thu, 28 Mar 2013 00:31:04 +0100 (CET) List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org To: alsa-devel@alsa-project.org Cc: jussi@sonarnerd.net, tiwai@suse.de, clemens@ladisch.de, Daniel Mack , demian@auraliti.com, ray@auraliti.com, andreas@akdesigninc.com List-Id: alsa-devel@alsa-project.org 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 --- 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