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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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directsoundsink: Implement SPDIF support for AC3.
Detect if the sound card supports SPDIF passthru of AC3 and add necessary code to support that like alsasink.
This commit is contained in:
Julien Moutte
parent
ab4e8ce9a9
commit
0e3c190d75
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@ -1,6 +1,7 @@
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/* GStreamer
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* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
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* Copyright (C) 2007 Pioneers of the Inevitable <songbird@songbirdnest.com>
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* Copyright (C) 2010 Fluendo S.A. <support@fluendo.com>
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*
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* gstdirectsoundsink.c:
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*
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@ -56,7 +57,19 @@
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#include <math.h>
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GST_DEBUG_CATEGORY_STATIC (directsoundsink_debug);
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#define GST_CAT_DEFAULT directsoundsink_debug
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/* elementfactory information */
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static const GstElementDetails gst_directsound_sink_details =
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GST_ELEMENT_DETAILS ("Direct Sound Audio Sink",
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"Sink/Audio",
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"Output to a sound card via Direct Sound",
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"Sebastien Moutte <sebastien@moutte.net>");
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static void gst_directsound_sink_base_init (gpointer g_class);
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static void gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass);
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static void gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
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GstDirectSoundSinkClass * g_class);
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static void gst_directsound_sink_finalise (GObject * object);
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static void gst_directsound_sink_set_property (GObject * object, guint prop_id,
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@ -75,6 +88,8 @@ static guint gst_directsound_sink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_directsound_sink_delay (GstAudioSink * asink);
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static void gst_directsound_sink_reset (GstAudioSink * asink);
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static GstCaps *gst_directsound_probe_supported_formats (GstDirectSoundSink *
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dsoundsink, const GstCaps * template_caps);
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/* interfaces */
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static void gst_directsound_sink_interfaces_init (GType type);
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@ -96,7 +111,8 @@ static GstStaticPadTemplate directsoundsink_sink_factory =
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
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"audio/x-iec958"));
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enum
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{
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@ -244,10 +260,7 @@ gst_directsound_sink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details_simple (element_class,
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"Direct Sound Audio Sink", "Sink/Audio",
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"Output to a sound card via Direct Sound",
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"Sebastien Moutte <sebastien@moutte.net>");
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gst_element_class_set_details (element_class, &gst_directsound_sink_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&directsoundsink_sink_factory));
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}
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@ -314,6 +327,7 @@ gst_directsound_sink_init (GstDirectSoundSink * dsoundsink,
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dsoundsink->tracks = g_list_append (dsoundsink->tracks, track);
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dsoundsink->pDS = NULL;
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dsoundsink->cached_caps = NULL;
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dsoundsink->pDSBSecondary = NULL;
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dsoundsink->current_circular_offset = 0;
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dsoundsink->buffer_size = DSBSIZE_MIN;
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@ -358,13 +372,35 @@ gst_directsound_sink_get_property (GObject * object,
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static GstCaps *
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gst_directsound_sink_getcaps (GstBaseSink * bsink)
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{
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GstDirectSoundSink *dsoundsink;
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GstElementClass *element_class;
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GstPadTemplate *pad_template;
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GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (bsink);
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GstCaps *caps;
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dsoundsink = GST_DIRECTSOUND_SINK (bsink);
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if (dsoundsink->pDS == NULL) {
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GST_DEBUG_OBJECT (dsoundsink, "device not open, using template caps");
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return NULL; /* base class will get template caps for us */
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}
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return
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gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
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(dsoundsink)));
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if (dsoundsink->cached_caps) {
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GST_DEBUG_OBJECT (dsoundsink, "Returning cached caps: %s",
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gst_caps_to_string (dsoundsink->cached_caps));
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return gst_caps_ref (dsoundsink->cached_caps);
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}
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element_class = GST_ELEMENT_GET_CLASS (dsoundsink);
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pad_template = gst_element_class_get_pad_template (element_class, "sink");
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g_return_val_if_fail (pad_template != NULL, NULL);
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caps = gst_directsound_probe_supported_formats (dsoundsink,
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gst_pad_template_get_caps (pad_template));
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if (caps) {
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dsoundsink->cached_caps = gst_caps_ref (caps);
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}
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GST_DEBUG_OBJECT (dsoundsink, "returning caps %s", gst_caps_to_string (caps));
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return caps;
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}
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static gboolean
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@ -400,29 +436,43 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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DSBUFFERDESC descSecondary;
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WAVEFORMATEX wfx;
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/*save number of bytes per sample */
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/*save number of bytes per sample and buffer format */
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dsoundsink->bytes_per_sample = spec->bytes_per_sample;
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dsoundsink->buffer_format = spec->format;
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/* fill the WAVEFORMATEX struture with spec params */
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/* fill the WAVEFORMATEX structure with spec params */
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memset (&wfx, 0, sizeof (wfx));
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wfx.cbSize = sizeof (wfx);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = spec->channels;
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wfx.nSamplesPerSec = spec->rate;
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wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels;
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wfx.nBlockAlign = spec->bytes_per_sample;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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if (spec->format != GST_IEC958) {
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wfx.cbSize = sizeof (wfx);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = spec->channels;
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wfx.nSamplesPerSec = spec->rate;
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wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels;
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wfx.nBlockAlign = spec->bytes_per_sample;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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/* Create directsound buffer with size based on our configured
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* buffer_size (which is 200 ms by default) */
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dsoundsink->buffer_size =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
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GST_MSECOND);
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/* Create directsound buffer with size based on our configured
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* buffer_size (which is 200 ms by default) */
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dsoundsink->buffer_size =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time,
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GST_MSECOND);
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spec->segsize =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
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GST_MSECOND);
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spec->segtotal = dsoundsink->buffer_size / spec->segsize;
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spec->segsize =
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gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time,
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GST_MSECOND);
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spec->segtotal = dsoundsink->buffer_size / spec->segsize;
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} else {
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wfx.cbSize = 0;
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wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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wfx.nChannels = 2;
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wfx.nSamplesPerSec = spec->rate;
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wfx.wBitsPerSample = 16;
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wfx.nBlockAlign = wfx.wBitsPerSample / 8 * wfx.nChannels;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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spec->segsize = 6144;
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spec->segtotal = 10;
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}
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// Make the final buffer size be an integer number of segments
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dsoundsink->buffer_size = spec->segsize * spec->segtotal;
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@ -430,15 +480,16 @@ gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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GST_INFO_OBJECT (dsoundsink,
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"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
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"WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
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"Size of dsound cirucular buffe=>%d\n", spec->channels, spec->rate,
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"Size of dsound circular buffer=>%d\n", spec->channels, spec->rate,
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spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample,
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wfx.nBlockAlign, wfx.nAvgBytesPerSec, dsoundsink->buffer_size);
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/* create a secondary directsound buffer */
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memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
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descSecondary.dwSize = sizeof (DSBUFFERDESC);
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descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 |
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DSBCAPS_GLOBALFOCUS | DSBCAPS_CTRLVOLUME;
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descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
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if (spec->format != GST_IEC958)
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descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME;
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descSecondary.dwBufferBytes = dsoundsink->buffer_size;
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descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx;
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@ -465,8 +516,10 @@ gst_directsound_sink_unprepare (GstAudioSink * asink)
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* release secondary DirectSound buffer */
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if (dsoundsink->pDSBSecondary)
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if (dsoundsink->pDSBSecondary) {
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IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
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dsoundsink->pDSBSecondary = NULL;
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}
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return TRUE;
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}
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@ -481,6 +534,9 @@ gst_directsound_sink_close (GstAudioSink * asink)
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/* release DirectSound object */
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g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE);
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IDirectSound_Release (dsoundsink->pDS);
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dsoundsink->pDS = NULL;
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gst_caps_replace (&dsoundsink->cached_caps, NULL);
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return TRUE;
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}
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@ -497,6 +553,10 @@ gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
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dsoundsink = GST_DIRECTSOUND_SINK (asink);
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/* Fix endianness */
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if (dsoundsink->buffer_format == GST_IEC958)
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_swab (data, data, length);
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GST_DSOUND_LOCK (dsoundsink);
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/* get current buffer status */
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@ -648,3 +708,64 @@ gst_directsound_sink_reset (GstAudioSink * asink)
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GST_DSOUND_UNLOCK (dsoundsink);
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}
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/*
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* gst_directsound_probe_supported_formats:
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*
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* Takes the template caps and returns the subset which is actually
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* supported by this device.
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*
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*/
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static GstCaps *
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gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
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const GstCaps * template_caps)
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{
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HRESULT hRes;
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DSBUFFERDESC descSecondary;
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WAVEFORMATEX wfx;
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GstCaps *caps;
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caps = gst_caps_copy (template_caps);
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/*
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* Check availability of digital output by trying to create an SPDIF buffer
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*/
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/* fill the WAVEFORMATEX structure with some standard AC3 over SPDIF params */
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memset (&wfx, 0, sizeof (wfx));
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wfx.cbSize = 0;
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wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
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wfx.nChannels = 2;
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wfx.nSamplesPerSec = 48000;
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wfx.wBitsPerSample = 16;
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wfx.nBlockAlign = 4;
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wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
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// create a secondary directsound buffer
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memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
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descSecondary.dwSize = sizeof (DSBUFFERDESC);
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descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
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descSecondary.dwBufferBytes = 6144;
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descSecondary.lpwfxFormat = &wfx;
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hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary,
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&dsoundsink->pDSBSecondary, NULL);
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if (FAILED (hRes)) {
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GST_INFO_OBJECT (dsoundsink, "AC3 passthrough not supported "
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"(IDirectSound_CreateSoundBuffer returned: %s)\n",
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DXGetErrorString9 (hRes));
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caps =
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gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
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} else {
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GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
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hRes = IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
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if (FAILED (hRes)) {
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GST_DEBUG_OBJECT (dsoundsink,
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"(IDirectSoundBuffer_Release returned: %s)\n",
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DXGetErrorString9 (hRes));
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}
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}
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return caps;
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}
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@ -1,6 +1,7 @@
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/* GStreamer
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* Copyright (C) 2005 Sebastien Moutte <sebastien@moutte.net>
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* Copyright (C) 2007 Pioneers of the Inevitable <songbird@songbirdnest.com>
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* Copyright (C) 2010 Fluendo S.A. <support@fluendo.com>
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*
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* gstdirectsoundsink.h:
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*
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@ -35,6 +36,9 @@
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#include <windows.h>
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#include <dxerr9.h>
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#include <dsound.h>
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#include <mmreg.h>
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#include <ks.h>
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#include <ksmedia.h>
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G_BEGIN_DECLS
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#define GST_TYPE_DIRECTSOUND_SINK (gst_directsound_sink_get_type())
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@ -72,10 +76,14 @@ struct _GstDirectSoundSink
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/* tracks list of our mixer interface implementation */
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GList *tracks;
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GstCaps *cached_caps;
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/* lock used to protect writes and resets */
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GMutex *dsound_lock;
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gboolean first_buffer_after_reset;
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GstBufferFormat buffer_format;
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};
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struct _GstDirectSoundSinkClass
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