static gboolean gst_directsound_sink_open (GstAudioSink * asink) { GstDirectSoundSink *dsoundsink; HRESULT hRes; dsoundsink = GST_DIRECTSOUND_SINK (asink); /* create and initialize a DirecSound object */ if (FAILED (hRes = DirectSoundCreate (NULL, &dsoundsink->pDS, NULL))) { GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ, ("gst_directsound_sink_open: DirectSoundCreate: %s", DXGetErrorString9 (hRes)), (NULL)); return FALSE; } if (FAILED (hRes = IDirectSound_SetCooperativeLevel (dsoundsink->pDS, GetDesktopWindow (), DSSCL_PRIORITY))) { GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ, ("gst_directsound_sink_open: IDirectSound_SetCooperativeLevel: %s", DXGetErrorString9 (hRes)), (NULL)); return FALSE; } return TRUE; }
static guint gst_directsound_sink_delay (GstAudioSink * asink) { GstDirectSoundSink *dsoundsink; HRESULT hRes; DWORD dwCurrentPlayCursor; DWORD dwBytesInQueue = 0; gint nNbSamplesInQueue = 0; DWORD dwStatus; dsoundsink = GST_DIRECTSOUND_SINK (asink); /* get current buffer status */ hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus); if (dwStatus & DSBSTATUS_PLAYING) { /*evaluate the number of samples in queue in the circular buffer */ hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary, &dwCurrentPlayCursor, NULL); if (hRes == S_OK) { if (dwCurrentPlayCursor < dsoundsink->current_circular_offset) dwBytesInQueue = dsoundsink->current_circular_offset - dwCurrentPlayCursor; else dwBytesInQueue = dsoundsink->current_circular_offset + (dsoundsink->buffer_size - dwCurrentPlayCursor); nNbSamplesInQueue = dwBytesInQueue / dsoundsink->bytes_per_sample; } } return nNbSamplesInQueue; }
static const GList * gst_directsound_sink_mixer_list_tracks (GstMixer * mixer) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer); return dsoundsink->tracks; }
static void gst_directsound_sink_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer); volumes[0] = dsoundsink->volume; }
static void gst_directsound_sink_finalize (GObject * object) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object); g_mutex_clear (&dsoundsink->dsound_lock); G_OBJECT_CLASS (parent_class)->finalize (object); }
static void gst_directsound_sink_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (mixer); if (volumes[0] != dsoundsink->volume) { dsoundsink->volume = volumes[0]; gst_directsound_sink_set_volume (dsoundsink); } }
static gboolean gst_directsound_sink_acceptcaps (GstPad * pad, GstCaps * caps) { GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (gst_pad_get_parent_element (pad)); GstCaps *pad_caps; GstStructure *st; gboolean ret = FALSE; GstRingBufferSpec spec = { 0 }; pad_caps = gst_pad_get_caps_reffed (pad); if (pad_caps) { gboolean cret = gst_caps_can_intersect (pad_caps, caps); gst_caps_unref (pad_caps); if (!cret) goto done; } /* If we've not got fixed caps, creating a stream might fail, so let's just * return from here with default acceptcaps behaviour */ if (!gst_caps_is_fixed (caps)) goto done; /* parse helper expects this set, so avoid nasty warning * will be set properly later on anyway */ spec.latency_time = GST_SECOND; if (!gst_ring_buffer_parse_caps (&spec, caps)) goto done; /* Make sure input is framed (one frame per buffer) and can be payloaded */ switch (spec.type) { case GST_BUFTYPE_AC3: case GST_BUFTYPE_DTS: { gboolean framed = FALSE, parsed = FALSE; st = gst_caps_get_structure (caps, 0); gst_structure_get_boolean (st, "framed", &framed); gst_structure_get_boolean (st, "parsed", &parsed); if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0) goto done; } default: break; } ret = TRUE; done: gst_object_unref (dsink); return ret; }
static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink) { GstDirectSoundSink *dsoundsink; dsoundsink = GST_DIRECTSOUND_SINK (asink); /* release secondary DirectSound buffer */ if (dsoundsink->pDSBSecondary) { IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary); dsoundsink->pDSBSecondary = NULL; } return TRUE; }
static void gst_directsound_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object); switch (prop_id) { case PROP_VOLUME: g_value_set_double (value, (double) sink->volume / 100.); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }
static void gst_directsound_sink_finalise (GObject * object) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (object); g_mutex_free (dsoundsink->dsound_lock); if (dsoundsink->tracks) { g_list_foreach (dsoundsink->tracks, (GFunc) g_object_unref, NULL); g_list_free (dsoundsink->tracks); dsoundsink->tracks = NULL; } G_OBJECT_CLASS (parent_class)->finalize (object); }
static gboolean gst_directsound_sink_close (GstAudioSink * asink) { GstDirectSoundSink *dsoundsink = NULL; dsoundsink = GST_DIRECTSOUND_SINK (asink); /* release DirectSound object */ g_return_val_if_fail (dsoundsink->pDS != NULL, FALSE); IDirectSound_Release (dsoundsink->pDS); dsoundsink->pDS = NULL; gst_caps_replace (&dsoundsink->cached_caps, NULL); return TRUE; }
static void gst_directsound_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstDirectSoundSink *sink = GST_DIRECTSOUND_SINK (object); switch (prop_id) { case PROP_VOLUME: gst_directsound_sink_set_volume (sink, g_value_get_double (value), TRUE); break; case PROP_MUTE: gst_directsound_sink_set_mute (sink, g_value_get_boolean (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } }
static GstCaps * gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter) { GstElementClass *element_class; GstPadTemplate *pad_template; GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (bsink); GstCaps *caps; if (dsoundsink->pDS == NULL) { GST_DEBUG_OBJECT (dsoundsink, "device not open, using template caps"); return NULL; /* base class will get template caps for us */ } if (dsoundsink->cached_caps) { caps = gst_caps_ref (dsoundsink->cached_caps); } else { element_class = GST_ELEMENT_GET_CLASS (dsoundsink); pad_template = gst_element_class_get_pad_template (element_class, "sink"); g_return_val_if_fail (pad_template != NULL, NULL); caps = gst_directsound_probe_supported_formats (dsoundsink, gst_pad_template_get_caps (pad_template)); if (caps) dsoundsink->cached_caps = gst_caps_ref (caps); } if (caps && filter) { GstCaps *tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = tmp; } if (caps) { gchar *caps_string = gst_caps_to_string (caps); GST_DEBUG_OBJECT (dsoundsink, "returning caps %s", caps_string); g_free (caps_string); } return caps; }
static void gst_directsound_sink_reset (GstAudioSink * asink) { GstDirectSoundSink *dsoundsink; LPVOID pLockedBuffer = NULL; DWORD dwSizeBuffer = 0; dsoundsink = GST_DIRECTSOUND_SINK (asink); GST_DSOUND_LOCK (dsoundsink); if (dsoundsink->pDSBSecondary) { /*stop playing */ HRESULT hRes = IDirectSoundBuffer_Stop (dsoundsink->pDSBSecondary); /*reset position */ hRes = IDirectSoundBuffer_SetCurrentPosition (dsoundsink->pDSBSecondary, 0); dsoundsink->current_circular_offset = 0; /*reset the buffer */ hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary, dsoundsink->current_circular_offset, dsoundsink->buffer_size, &pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L); if (SUCCEEDED (hRes)) { memset (pLockedBuffer, 0, dwSizeBuffer); hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer, dwSizeBuffer, NULL, 0); } } dsoundsink->first_buffer_after_reset = TRUE; GST_DSOUND_UNLOCK (dsoundsink); }
static guint gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length) { GstDirectSoundSink *dsoundsink; DWORD dwStatus; HRESULT hRes; LPVOID pLockedBuffer1 = NULL, pLockedBuffer2 = NULL; DWORD dwSizeBuffer1, dwSizeBuffer2; DWORD dwCurrentPlayCursor; dsoundsink = GST_DIRECTSOUND_SINK (asink); /* Fix endianness */ if (dsoundsink->buffer_format == GST_IEC958) _swab (data, data, length); GST_DSOUND_LOCK (dsoundsink); /* get current buffer status */ hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus); /* get current play cursor position */ hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary, &dwCurrentPlayCursor, NULL); if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) { DWORD dwFreeBufferSize; calculate_freesize: /* calculate the free size of the circular buffer */ if (dwCurrentPlayCursor < dsoundsink->current_circular_offset) dwFreeBufferSize = dsoundsink->buffer_size - (dsoundsink->current_circular_offset - dwCurrentPlayCursor); else dwFreeBufferSize = dwCurrentPlayCursor - dsoundsink->current_circular_offset; if (length >= dwFreeBufferSize) { Sleep (100); hRes = IDirectSoundBuffer_GetCurrentPosition (dsoundsink->pDSBSecondary, &dwCurrentPlayCursor, NULL); hRes = IDirectSoundBuffer_GetStatus (dsoundsink->pDSBSecondary, &dwStatus); if (SUCCEEDED (hRes) && (dwStatus & DSBSTATUS_PLAYING)) goto calculate_freesize; else { dsoundsink->first_buffer_after_reset = FALSE; GST_DSOUND_UNLOCK (dsoundsink); return 0; } } } if (dwStatus & DSBSTATUS_BUFFERLOST) { hRes = IDirectSoundBuffer_Restore (dsoundsink->pDSBSecondary); /*need a loop waiting the buffer is restored?? */ dsoundsink->current_circular_offset = 0; } hRes = IDirectSoundBuffer_Lock (dsoundsink->pDSBSecondary, dsoundsink->current_circular_offset, length, &pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L); if (SUCCEEDED (hRes)) { // Write to pointers without reordering. memcpy (pLockedBuffer1, data, dwSizeBuffer1); if (pLockedBuffer2 != NULL) memcpy (pLockedBuffer2, (LPBYTE) data + dwSizeBuffer1, dwSizeBuffer2); // Update where the buffer will lock (for next time) dsoundsink->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2; dsoundsink->current_circular_offset %= dsoundsink->buffer_size; /* Circular buffer */ hRes = IDirectSoundBuffer_Unlock (dsoundsink->pDSBSecondary, pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2); } /* if the buffer was not in playing state yet, call play on the buffer except if this buffer is the fist after a reset (base class call reset and write a buffer when setting the sink to pause) */ if (!(dwStatus & DSBSTATUS_PLAYING) && dsoundsink->first_buffer_after_reset == FALSE) { hRes = IDirectSoundBuffer_Play (dsoundsink->pDSBSecondary, 0, 0, DSBPLAY_LOOPING); } dsoundsink->first_buffer_after_reset = FALSE; GST_DSOUND_UNLOCK (dsoundsink); return length; }
static gboolean gst_directsound_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) { GstDirectSoundSink *dsoundsink = GST_DIRECTSOUND_SINK (asink); HRESULT hRes; DSBUFFERDESC descSecondary; WAVEFORMATEX wfx; /*save number of bytes per sample and buffer format */ dsoundsink->bytes_per_sample = spec->bytes_per_sample; dsoundsink->buffer_format = spec->format; /* fill the WAVEFORMATEX structure with spec params */ memset (&wfx, 0, sizeof (wfx)); if (spec->format != GST_IEC958) { wfx.cbSize = sizeof (wfx); wfx.wFormatTag = WAVE_FORMAT_PCM; wfx.nChannels = spec->channels; wfx.nSamplesPerSec = spec->rate; wfx.wBitsPerSample = (spec->bytes_per_sample * 8) / wfx.nChannels; wfx.nBlockAlign = spec->bytes_per_sample; wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; /* Create directsound buffer with size based on our configured * buffer_size (which is 200 ms by default) */ dsoundsink->buffer_size = gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->buffer_time, GST_MSECOND); /* Make sure we make those numbers multiple of our sample size in bytes */ dsoundsink->buffer_size += dsoundsink->buffer_size % spec->bytes_per_sample; spec->segsize = gst_util_uint64_scale_int (wfx.nAvgBytesPerSec, spec->latency_time, GST_MSECOND); spec->segsize += spec->segsize % spec->bytes_per_sample; spec->segtotal = dsoundsink->buffer_size / spec->segsize; } else { wfx.cbSize = 0; wfx.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF; wfx.nChannels = 2; wfx.nSamplesPerSec = spec->rate; wfx.wBitsPerSample = 16; wfx.nBlockAlign = wfx.wBitsPerSample / 8 * wfx.nChannels; wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign; spec->segsize = 6144; spec->segtotal = 10; } // Make the final buffer size be an integer number of segments dsoundsink->buffer_size = spec->segsize * spec->segtotal; GST_INFO_OBJECT (dsoundsink, "GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n" "WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n" "Size of dsound circular buffer=>%d\n", spec->channels, spec->rate, spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nBlockAlign, wfx.nAvgBytesPerSec, dsoundsink->buffer_size); /* create a secondary directsound buffer */ memset (&descSecondary, 0, sizeof (DSBUFFERDESC)); descSecondary.dwSize = sizeof (DSBUFFERDESC); descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS; if (spec->format != GST_IEC958) descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME; descSecondary.dwBufferBytes = dsoundsink->buffer_size; descSecondary.lpwfxFormat = (WAVEFORMATEX *) & wfx; hRes = IDirectSound_CreateSoundBuffer (dsoundsink->pDS, &descSecondary, &dsoundsink->pDSBSecondary, NULL); if (FAILED (hRes)) { GST_ELEMENT_ERROR (dsoundsink, RESOURCE, OPEN_READ, ("gst_directsound_sink_prepare: IDirectSound_CreateSoundBuffer: %s", DXGetErrorString9 (hRes)), (NULL)); return FALSE; } gst_directsound_sink_set_volume (dsoundsink); return TRUE; }
static gboolean gst_directsound_sink_acceptcaps (GstBaseSink * sink, GstQuery * query) { GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (sink); GstPad *pad; GstCaps *caps; GstCaps *pad_caps; GstStructure *st; gboolean ret = FALSE; GstAudioRingBufferSpec spec = { 0 }; if (G_UNLIKELY (dsink == NULL)) return FALSE; pad = sink->sinkpad; gst_query_parse_accept_caps (query, &caps); GST_DEBUG_OBJECT (pad, "caps %" GST_PTR_FORMAT, caps); pad_caps = gst_pad_query_caps (pad, NULL); if (pad_caps) { gboolean cret = gst_caps_is_subset (caps, pad_caps); gst_caps_unref (pad_caps); if (!cret) { GST_DEBUG_OBJECT (dsink, "Caps are not a subset of the pad caps, not accepting caps"); goto done; } } /* If we've not got fixed caps, creating a stream might fail, so let's just * return from here with default acceptcaps behaviour */ if (!gst_caps_is_fixed (caps)) { GST_DEBUG_OBJECT (dsink, "Caps are not fixed, not accepting caps"); goto done; } spec.latency_time = GST_SECOND; if (!gst_audio_ring_buffer_parse_caps (&spec, caps)) { GST_DEBUG_OBJECT (dsink, "Failed to parse caps, not accepting"); goto done; } /* Make sure input is framed (one frame per buffer) and can be payloaded */ switch (spec.type) { case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3: case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS: { gboolean framed = FALSE, parsed = FALSE; st = gst_caps_get_structure (caps, 0); gst_structure_get_boolean (st, "framed", &framed); gst_structure_get_boolean (st, "parsed", &parsed); if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0) { GST_DEBUG_OBJECT (dsink, "Wrong AC3/DTS caps, not accepting"); goto done; } } default: break; } ret = TRUE; GST_DEBUG_OBJECT (dsink, "Accepting caps"); done: gst_query_set_accept_caps_result (query, ret); return TRUE; }