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authorMax Kellermann <max@duempel.org>2012-03-21 10:36:19 +0100
committerMax Kellermann <max@duempel.org>2012-03-22 01:30:16 +0100
commit167242fec045a104e91ca142fb3bd38af8cb78bb (patch)
tree52d67f865c2b63229cb2703f0a72c8fb64106b0f
parent81208d78acb260c7c2e6debc765042dd5e98f862 (diff)
downloadmpd-167242fec045a104e91ca142fb3bd38af8cb78bb.tar.gz
mpd-167242fec045a104e91ca142fb3bd38af8cb78bb.tar.xz
mpd-167242fec045a104e91ca142fb3bd38af8cb78bb.zip
output/alsa: add option to enable DSD over USB
Diffstat (limited to '')
-rw-r--r--NEWS1
-rw-r--r--doc/user.xml17
-rw-r--r--src/output/alsa_output_plugin.c55
3 files changed, 72 insertions, 1 deletions
diff --git a/NEWS b/NEWS
index 9343f3f77..d576787df 100644
--- a/NEWS
+++ b/NEWS
@@ -17,6 +17,7 @@ ver 0.17 (2011/??/??)
- oggflac: delete this obsolete plugin
- dsdiff: new decoder plugin
* output:
+ - alsa: support DSD-over-USB (dCS suggested standard)
- httpd: support for streaming to a DLNA client
- openal: improve buffer cancellation
- osx: allow user to specify other audio devices
diff --git a/doc/user.xml b/doc/user.xml
index 427e561c5..6c0547f0c 100644
--- a/doc/user.xml
+++ b/doc/user.xml
@@ -1119,6 +1119,23 @@ systemctl start mpd.socket</programlisting>
bit, floating point, ...).
</entry>
</row>
+ <row>
+ <entry>
+ <varname>dsd_usb</varname>
+ <parameter>yes|no</parameter>
+ </entry>
+ <entry>
+ If set to <parameter>yes</parameter>, then DSD over
+ USB according to the <ulink
+ url="http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf">dCS
+ suggested standard</ulink> is enabled. This wrapsa
+ DSD samples in fake 24 bit PCM, and is understood by
+ some DSD capable products, but may be harmful to
+ other hardware. Therefore, the default is
+ <parameter>no</parameter> and you can enable the
+ option at your own risk.
+ </entry>
+ </row>
</tbody>
</tgroup>
</informaltable>
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index 49b11c9b2..d0b0c4a5e 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/alsa_output_plugin.c
@@ -55,6 +55,14 @@ struct alsa_data {
/** use memory mapped I/O? */
bool use_mmap;
+ /**
+ * Enable DSD over USB according to the dCS suggested
+ * standard?
+ *
+ * @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
+ */
+ bool dsd_usb;
+
/** libasound's buffer_time setting (in microseconds) */
unsigned int buffer_time;
@@ -128,6 +136,8 @@ alsa_configure(struct alsa_data *ad, const struct config_param *param)
ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
+ ad->dsd_usb = config_get_block_bool(param, "dsd_usb", false);
+
ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
MPD_ALSA_BUFFER_TIME_US);
ad->period_time = config_get_block_unsigned(param, "period_time", 0);
@@ -576,6 +586,49 @@ error:
}
static bool
+alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+ GError **error_r)
+{
+ assert(ad->dsd_usb);
+ assert(audio_format->format == SAMPLE_FORMAT_DSD);
+
+ /* pass 24 bit to alsa_setup() */
+
+ audio_format->format = SAMPLE_FORMAT_S24_P32;
+ audio_format->sample_rate /= 2;
+
+ const struct audio_format check = *audio_format;
+
+ if (!alsa_setup(ad, audio_format, error_r))
+ return false;
+
+ if (!audio_format_equals(audio_format, &check)) {
+ /* no bit-perfect playback, which is required
+ for DSD over USB */
+ g_set_error(error_r, alsa_output_quark(), 0,
+ "Failed to configure DSD-over-USB on ALSA device \"%s\"",
+ alsa_device(ad));
+ return false;
+ }
+
+ /* the ALSA device has accepted 24 bit playback,
+ return DSD_OVER_USB to the caller */
+
+ audio_format->format = SAMPLE_FORMAT_DSD_OVER_USB;
+ return true;
+}
+
+static bool
+alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+ GError **error_r)
+{
+ if (ad->dsd_usb && audio_format->format == SAMPLE_FORMAT_DSD)
+ return alsa_setup_dsd(ad, audio_format, error_r);
+
+ return alsa_setup(ad, audio_format, error_r);
+}
+
+static bool
alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
{
struct alsa_data *ad = (struct alsa_data *)ao;
@@ -591,7 +644,7 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
return false;
}
- success = alsa_setup(ad, audio_format, error);
+ success = alsa_setup_or_dsd(ad, audio_format, error);
if (!success) {
snd_pcm_close(ad->pcm);
return false;