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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-18 14:26:43 +00:00
directsoundsink: add support for ac-3 over spdif
This commit is contained in:
parent
94e54887fb
commit
7d64e16b30
1 changed files with 150 additions and 11 deletions
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@ -55,6 +55,7 @@
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#include <gst/base/gstbasesink.h>
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#include <gst/audio/streamvolume.h>
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#include "gstdirectsoundsink.h"
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#include <gst/audio/gstaudioiec61937.h>
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#include <math.h>
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@ -79,6 +80,8 @@ static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
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static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink,
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GstCaps * filter);
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static GstBuffer *gst_directsound_sink_payload (GstAudioBaseSink * sink,
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GstBuffer * buf);
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static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink);
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@ -90,6 +93,7 @@ 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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static gboolean gst_directsound_sink_query (GstBaseSink * pad, GstQuery * query);
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static void gst_directsound_sink_set_volume (GstDirectSoundSink * sink,
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gdouble volume, gboolean store);
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@ -110,7 +114,7 @@ static GstStaticPadTemplate directsoundsink_sink_factory =
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"format = (string) S8, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
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"audio/x-iec958"));
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"audio/x-ac3, framed = (boolean) true;"));
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enum
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{
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@ -136,6 +140,7 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
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GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
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GstAudioBaseSinkClass *gstaudiobasesink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
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@ -150,6 +155,12 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
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gstbasesink_class->get_caps =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
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gstbasesink_class->query =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_query);
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gstaudiobasesink_class->payload =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_payload);
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gstaudiosink_class->prepare =
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GST_DEBUG_FUNCPTR (gst_directsound_sink_prepare);
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gstaudiosink_class->unprepare =
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@ -274,6 +285,82 @@ gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
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return caps;
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}
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static gboolean
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gst_directsound_sink_acceptcaps (GstBaseSink * sink, GstQuery * query)
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{
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GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (sink);
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GstAudioRingBuffer *rbuf = GST_AUDIO_BASE_SINK (dsink)->ringbuffer;
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GstPad *pad;
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GstCaps *caps;
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GstCaps *pad_caps;
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GstStructure *st;
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gboolean ret = FALSE;
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GstAudioRingBufferSpec spec = { 0 };
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if (G_UNLIKELY (dsink == NULL))
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return FALSE;
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pad = sink->sinkpad;
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gst_query_parse_accept_caps (query, &caps);
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GST_DEBUG_OBJECT (pad, "caps %" GST_PTR_FORMAT, caps);
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pad_caps = gst_pad_query_caps (pad, NULL);
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if (pad_caps) {
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ret = gst_caps_can_intersect (pad_caps, caps);
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gst_caps_unref (pad_caps);
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if (!ret)
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goto done;
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}
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/* If we've not got fixed caps, creating a stream might fail, so let's just
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* return from here with default acceptcaps behaviour */
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if (!gst_caps_is_fixed (caps))
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goto done;
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if (!gst_audio_ring_buffer_parse_caps (&rbuf->spec, caps))
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goto done;
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/* Make sure input is framed (one frame per buffer) and can be payloaded */
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switch (rbuf->spec.type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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{
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gboolean framed = FALSE, parsed = FALSE;
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st = gst_caps_get_structure (caps, 0);
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gst_structure_get_boolean (st, "framed", &framed);
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gst_structure_get_boolean (st, "parsed", &parsed);
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if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
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goto done;
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}
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default:{
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}
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}
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ret = TRUE;
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done:
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gst_object_unref (dsink);
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gst_query_set_accept_caps_result (query, ret);
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return TRUE;
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}
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static gboolean
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gst_directsound_sink_query (GstBaseSink * sink, GstQuery * query)
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{
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gboolean res = TRUE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_ACCEPT_CAPS:
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res = gst_directsound_sink_acceptcaps (sink, query);
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break;
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default:
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res = FALSE;
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}
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return res;
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}
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static gboolean
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gst_directsound_sink_open (GstAudioSink * asink)
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{
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@ -301,6 +388,14 @@ gst_directsound_sink_open (GstAudioSink * asink)
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return TRUE;
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}
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static boolean
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gst_directsound_sink_is_spdif_format (GstDirectSoundSink * dsoundsink)
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{
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GstAudioRingBufferFormatType type;
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type = GST_AUDIO_BASE_SINK (dsoundsink)->ringbuffer->spec.type;
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return type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3;
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}
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static gboolean
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gst_directsound_sink_prepare (GstAudioSink * asink,
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GstAudioRingBufferSpec * spec)
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@ -318,7 +413,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
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/* fill the WAVEFORMATEX structure with spec params */
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memset (&wfx, 0, sizeof (wfx));
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if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958) {
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if (gst_directsound_sink_is_spdif_format (dsoundsink)) {
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wfx.cbSize = sizeof (wfx);
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wfx.wFormatTag = WAVE_FORMAT_PCM;
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wfx.nChannels = spec->info.channels;
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@ -361,7 +456,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
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dsoundsink->buffer_size = spec->segsize * spec->segtotal;
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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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"GstAudioRingBufferSpec->channels: %d, GstAudioRingBufferSpec->rate: %d, GstAudioRingBufferSpec->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 circular buffer=>%d\n", spec->info.channels,
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spec->info.rate, spec->info.bpf, wfx.nSamplesPerSec, wfx.wBitsPerSample,
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@ -371,7 +466,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
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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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if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958)
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if (!gst_directsound_sink_is_spdif_format (dsoundsink))
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descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME;
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descSecondary.dwBufferBytes = dsoundsink->buffer_size;
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@ -436,10 +531,6 @@ 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->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_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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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_empty_simple ("audio/x-iec958"));
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caps = gst_caps_subtract (caps, gst_caps_new_empty_simple ("audio/x-ac3"));
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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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}
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}
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#else
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caps = gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
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caps = gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-ac3", NULL));
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#endif
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return caps;
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}
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static GstBuffer *
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gst_directsound_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
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{
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switch (sink->ringbuffer->spec.type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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{
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gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
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GstBuffer *out;
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GstMapInfo infobuf, infoout;
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gboolean success;
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if (framesize <= 0)
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return NULL;
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out = gst_buffer_new_and_alloc (framesize);
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if (!gst_buffer_map (buf, &infobuf, GST_MAP_READWRITE))
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{
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gst_buffer_unref (out);
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return NULL;
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}
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if (!gst_buffer_map (out, &infoout, GST_MAP_READWRITE))
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{
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gst_buffer_unmap (buf, &infobuf);
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gst_buffer_unref (out);
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return NULL;
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}
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success = gst_audio_iec61937_payload (infobuf.data, infobuf.size,
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infoout.data, infoout.size, &sink->ringbuffer->spec);
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if (!success) {
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gst_buffer_unmap (out, &infoout);
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gst_buffer_unmap (buf, &infobuf);
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gst_buffer_unref (out);
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return NULL;
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}
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gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_ALL, 0, -1);
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/* Fix endianness */
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_swab ((gchar *) infoout.data, (gchar *) infoout.data, infobuf.size);
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gst_buffer_unmap (out, &infoout);
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gst_buffer_unmap (buf, &infobuf);
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return out;
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}
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default:
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return gst_buffer_ref (buf);
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}
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}
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static void
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gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink,
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gdouble dvolume, gboolean store)
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