void gst_decklink_audio_sink_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); switch (property_id) { case PROP_DEVICE_NUMBER: self->device_number = g_value_get_int (value); break; case PROP_ALIGNMENT_THRESHOLD: GST_OBJECT_LOCK (self); gst_audio_stream_align_set_alignment_threshold (self->stream_align, g_value_get_uint64 (value)); GST_OBJECT_UNLOCK (self); break; case PROP_DISCONT_WAIT: GST_OBJECT_LOCK (self); gst_audio_stream_align_set_discont_wait (self->stream_align, g_value_get_uint64 (value)); GST_OBJECT_UNLOCK (self); break; case PROP_BUFFER_TIME: GST_OBJECT_LOCK (self); self->buffer_time = g_value_get_uint64 (value) * 1000; GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } }
static GstCaps * gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GstCaps *caps; if ((caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (bsink)))) return caps; caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink)); GST_OBJECT_LOCK (self); if (self->output && self->output->attributes) { int64_t max_channels = 0; HRESULT ret; GstStructure *s; GValue arr = G_VALUE_INIT; GValue v = G_VALUE_INIT; ret = self->output->attributes->GetInt (BMDDeckLinkMaximumAudioChannels, &max_channels); /* 2 should always be supported */ if (ret != S_OK) { max_channels = 2; } caps = gst_caps_make_writable (caps); s = gst_caps_get_structure (caps, 0); g_value_init (&arr, GST_TYPE_LIST); g_value_init (&v, G_TYPE_INT); if (max_channels >= 16) { g_value_set_int (&v, 16); gst_value_list_append_value (&arr, &v); } if (max_channels >= 8) { g_value_set_int (&v, 8); gst_value_list_append_value (&arr, &v); } g_value_set_int (&v, 2); gst_value_list_append_value (&arr, &v); gst_structure_set_value (s, "channels", &arr); g_value_unset (&v); g_value_unset (&arr); } GST_OBJECT_UNLOCK (self); if (filter) { GstCaps *intersection = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = intersection; } return caps; }
static GstClock * gst_decklink_audio_sink_provide_clock (GstElement * element) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element); if (!self->output) return NULL; return GST_CLOCK_CAST (gst_object_ref (self->output->clock)); }
void gst_decklink_audio_sink_finalize (GObject * object) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); if (self->stream_align) { gst_audio_stream_align_free (self->stream_align); self->stream_align = NULL; } G_OBJECT_CLASS (parent_class)->finalize (object); }
void gst_decklink_audio_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); switch (property_id) { case PROP_DEVICE_NUMBER: g_value_set_int (value, self->device_number); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } }
static gboolean gst_decklink_audio_sink_set_caps (GstBaseSink * bsink, GstCaps * caps) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); HRESULT ret; BMDAudioSampleType sample_depth; GstAudioInfo info; GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps); if (!gst_audio_info_from_caps (&info, caps)) return FALSE; if (self->output->audio_enabled && (self->info.finfo->format != info.finfo->format || self->info.channels != info.channels)) { GST_ERROR_OBJECT (self, "Reconfiguration not supported"); return FALSE; } else if (self->output->audio_enabled) { return TRUE; } if (info.finfo->format == GST_AUDIO_FORMAT_S16LE) { sample_depth = bmdAudioSampleType16bitInteger; } else { sample_depth = bmdAudioSampleType32bitInteger; } ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz, sample_depth, info.channels, bmdAudioOutputStreamContinuous); if (ret != S_OK) { GST_WARNING_OBJECT (self, "Failed to enable audio output 0x%08lx", (unsigned long) ret); return FALSE; } self->output->audio_enabled = TRUE; self->info = info; // Create a new resampler as needed if (self->resampler) gst_audio_resampler_free (self->resampler); self->resampler = NULL; return TRUE; }
static gboolean gst_decklink_audio_sink_open (GstBaseSink * bsink) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GST_DEBUG_OBJECT (self, "Starting"); self->output = gst_decklink_acquire_nth_output (self->device_number, GST_ELEMENT_CAST (self), TRUE); if (!self->output) { GST_ERROR_OBJECT (self, "Failed to acquire output"); return FALSE; } g_object_notify (G_OBJECT (self), "hw-serial-number"); return TRUE; }
void gst_decklink_audio_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); switch (property_id) { case PROP_DEVICE_NUMBER: g_value_set_int (value, self->device_number); break; case PROP_HW_SERIAL_NUMBER: if (self->output) g_value_set_string (value, self->output->hw_serial_number); else g_value_set_string (value, NULL); break; case PROP_ALIGNMENT_THRESHOLD: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, gst_audio_stream_align_get_alignment_threshold (self->stream_align)); GST_OBJECT_UNLOCK (self); break; case PROP_DISCONT_WAIT: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, gst_audio_stream_align_get_discont_wait (self->stream_align)); GST_OBJECT_UNLOCK (self); break; case PROP_BUFFER_TIME: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, self->buffer_time / 1000); GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } }
static gboolean gst_decklink_audio_sink_close (GstBaseSink * bsink) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GST_DEBUG_OBJECT (self, "Closing"); if (self->output) { g_mutex_lock (&self->output->lock); self->output->mode = NULL; self->output->audio_enabled = FALSE; if (self->output->start_scheduled_playback && self->output->videosink) self->output->start_scheduled_playback (self->output->videosink); g_mutex_unlock (&self->output->lock); self->output->output->DisableAudioOutput (); gst_decklink_release_nth_output (self->device_number, GST_ELEMENT_CAST (self), TRUE); self->output = NULL; } return TRUE; }
static GstStateChangeReturn gst_decklink_audio_sink_change_state (GstElement * element, GstStateChange transition) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: GST_OBJECT_LOCK (self); gst_audio_stream_align_mark_discont (self->stream_align); GST_OBJECT_UNLOCK (self); g_mutex_lock (&self->output->lock); if (self->output->start_scheduled_playback) self->output->start_scheduled_playback (self->output->videosink); g_mutex_unlock (&self->output->lock); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_decklink_audio_sink_stop (self); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) return ret; switch (transition) { default: break; } return ret; }
static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) { const GstSegment *new_segment; gst_event_parse_segment (event, &new_segment); if (ABS (new_segment->rate) != 1.0) { guint out_rate = self->info.rate / ABS (new_segment->rate); if (self->resampler && (self->resampler_out_rate != out_rate || self->resampler_in_rate != (guint) self->info.rate)) gst_audio_resampler_update (self->resampler, self->info.rate, out_rate, NULL); else if (!self->resampler) self->resampler = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_LINEAR, GST_AUDIO_RESAMPLER_FLAG_NONE, self->info.finfo->format, self->info.channels, self->info.rate, out_rate, NULL); self->resampler_in_rate = self->info.rate; self->resampler_out_rate = out_rate; } else if (self->resampler) { gst_audio_resampler_free (self->resampler); self->resampler = NULL; } if (new_segment->rate < 0) gst_audio_stream_align_set_rate (self->stream_align, -48000); } return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event); }
static GstFlowReturn gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GstDecklinkVideoSink *video_sink; GstFlowReturn flow_ret; HRESULT ret; GstClockTime timestamp, duration; GstClockTime running_time, running_time_duration; GstClockTime schedule_time, schedule_time_duration; GstClockTime latency, render_delay; GstClockTimeDiff ts_offset; GstMapInfo map_info; const guint8 *data; gsize len, written_all; gboolean discont; GST_DEBUG_OBJECT (self, "Rendering buffer %p", buffer); // FIXME: Handle no timestamps if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { return GST_FLOW_ERROR; } if (GST_BASE_SINK_CAST (self)->flushing) { return GST_FLOW_FLUSHING; } // If we're called before output is actually started, start pre-rolling if (!self->output->started) { self->output->output->BeginAudioPreroll (); } video_sink = GST_DECKLINK_VIDEO_SINK (gst_object_ref (self->output->videosink)); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); discont = gst_audio_stream_align_process (self->stream_align, GST_BUFFER_IS_DISCONT (buffer), timestamp, gst_buffer_get_size (buffer) / self->info.bpf, ×tamp, &duration, NULL); if (discont && self->resampler) gst_audio_resampler_reset (self->resampler); if (GST_BASE_SINK_CAST (self)->segment.rate < 0.0) { GstMapInfo out_map; gint out_frames = gst_buffer_get_size (buffer) / self->info.bpf; buffer = gst_buffer_make_writable (gst_buffer_ref (buffer)); gst_buffer_map (buffer, &out_map, GST_MAP_READWRITE); if (self->info.finfo->format == GST_AUDIO_FORMAT_S16) { gint16 *swap_data = (gint16 *) out_map.data; gint16 *swap_data_end = swap_data + (out_frames - 1) * self->info.channels; gint16 swap_tmp[16]; while (out_frames > 0) { memcpy (&swap_tmp, swap_data, self->info.bpf); memcpy (swap_data, swap_data_end, self->info.bpf); memcpy (swap_data_end, &swap_tmp, self->info.bpf); swap_data += self->info.channels; swap_data_end -= self->info.channels; out_frames -= 2; } } else { gint32 *swap_data = (gint32 *) out_map.data; gint32 *swap_data_end = swap_data + (out_frames - 1) * self->info.channels; gint32 swap_tmp[16]; while (out_frames > 0) { memcpy (&swap_tmp, swap_data, self->info.bpf); memcpy (swap_data, swap_data_end, self->info.bpf); memcpy (swap_data_end, &swap_tmp, self->info.bpf); swap_data += self->info.channels; swap_data_end -= self->info.channels; out_frames -= 2; } } gst_buffer_unmap (buffer, &out_map); } else { gst_buffer_ref (buffer); } if (self->resampler) { gint in_frames = gst_buffer_get_size (buffer) / self->info.bpf; gint out_frames = gst_audio_resampler_get_out_frames (self->resampler, in_frames); GstBuffer *out_buf = gst_buffer_new_and_alloc (out_frames * self->info.bpf); GstMapInfo out_map; gst_buffer_map (buffer, &map_info, GST_MAP_READ); gst_buffer_map (out_buf, &out_map, GST_MAP_READWRITE); gst_audio_resampler_resample (self->resampler, (gpointer *) & map_info.data, in_frames, (gpointer *) & out_map.data, out_frames); gst_buffer_unmap (out_buf, &out_map); gst_buffer_unmap (buffer, &map_info); buffer = out_buf; } gst_buffer_map (buffer, &map_info, GST_MAP_READ); data = map_info.data; len = map_info.size / self->info.bpf; written_all = 0; do { GstClockTime timestamp_now = timestamp + gst_util_uint64_scale (written_all, GST_SECOND, self->info.rate); guint32 buffered_samples; GstClockTime buffered_time; guint32 written = 0; GstClock *clock; GstClockTime clock_ahead; if (GST_BASE_SINK_CAST (self)->flushing) { flow_ret = GST_FLOW_FLUSHING; break; } running_time = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp_now); running_time_duration = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp_now + duration) - running_time; /* See gst_base_sink_adjust_time() */ latency = gst_base_sink_get_latency (bsink); render_delay = gst_base_sink_get_render_delay (bsink); ts_offset = gst_base_sink_get_ts_offset (bsink); running_time += latency; if (ts_offset < 0) { ts_offset = -ts_offset; if ((GstClockTime) ts_offset < running_time) running_time -= ts_offset; else running_time = 0; } else { running_time += ts_offset; } if (running_time > render_delay) running_time -= render_delay; else running_time = 0; clock = gst_element_get_clock (GST_ELEMENT_CAST (self)); clock_ahead = 0; if (clock) { GstClockTime clock_now = gst_clock_get_time (clock); GstClockTime base_time = gst_element_get_base_time (GST_ELEMENT_CAST (self)); gst_object_unref (clock); clock = NULL; if (clock_now != GST_CLOCK_TIME_NONE && base_time != GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (self, "Clock time %" GST_TIME_FORMAT ", base time %" GST_TIME_FORMAT ", target running time %" GST_TIME_FORMAT, GST_TIME_ARGS (clock_now), GST_TIME_ARGS (base_time), GST_TIME_ARGS (running_time)); if (clock_now > base_time) clock_now -= base_time; else clock_now = 0; if (clock_now < running_time) clock_ahead = running_time - clock_now; } } GST_DEBUG_OBJECT (self, "Ahead %" GST_TIME_FORMAT " of the clock running time", GST_TIME_ARGS (clock_ahead)); if (self->output-> output->GetBufferedAudioSampleFrameCount (&buffered_samples) != S_OK) buffered_samples = 0; buffered_time = gst_util_uint64_scale (buffered_samples, GST_SECOND, self->info.rate); buffered_time /= ABS (GST_BASE_SINK_CAST (self)->segment.rate); GST_DEBUG_OBJECT (self, "Buffered %" GST_TIME_FORMAT " in the driver (%u samples)", GST_TIME_ARGS (buffered_time), buffered_samples); // We start waiting once we have more than buffer-time buffered if (buffered_time > self->buffer_time || clock_ahead > self->buffer_time) { GstClockReturn clock_ret; GstClockTime wait_time = running_time; GST_DEBUG_OBJECT (self, "Buffered enough, wait for preroll or the clock or flushing"); if (wait_time < self->buffer_time) wait_time = 0; else wait_time -= self->buffer_time; flow_ret = gst_base_sink_do_preroll (GST_BASE_SINK_CAST (self), GST_MINI_OBJECT_CAST (buffer)); if (flow_ret != GST_FLOW_OK) break; clock_ret = gst_base_sink_wait_clock (GST_BASE_SINK_CAST (self), wait_time, NULL); if (GST_BASE_SINK_CAST (self)->flushing) { flow_ret = GST_FLOW_FLUSHING; break; } // Rerun the whole loop again if (clock_ret == GST_CLOCK_UNSCHEDULED) continue; } schedule_time = running_time; schedule_time_duration = running_time_duration; gst_decklink_video_sink_convert_to_internal_clock (video_sink, &schedule_time, &schedule_time_duration); GST_LOG_OBJECT (self, "Scheduling audio samples at %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT, GST_TIME_ARGS (schedule_time), GST_TIME_ARGS (schedule_time_duration)); ret = self->output->output->ScheduleAudioSamples ((void *) data, len, schedule_time, GST_SECOND, &written); if (ret != S_OK) { bool is_running = true; self->output->output->IsScheduledPlaybackRunning (&is_running); if (is_running && !GST_BASE_SINK_CAST (self)->flushing && self->output->started) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to schedule frame: 0x%08lx", (unsigned long) ret)); flow_ret = GST_FLOW_ERROR; break; } else { // Ignore the error and go out of the loop here, we're shutting down // or are not started yet and there's nothing we can do at this point GST_INFO_OBJECT (self, "Ignoring scheduling error 0x%08x because we're not started yet" " or not anymore", (guint) ret); flow_ret = GST_FLOW_OK; break; } } len -= written; data += written * self->info.bpf; if (self->resampler) written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate); else written_all += written; flow_ret = GST_FLOW_OK; } while (len > 0); gst_buffer_unmap (buffer, &map_info); gst_buffer_unref (buffer); GST_DEBUG_OBJECT (self, "Returning %s", gst_flow_get_name (flow_ret)); return flow_ret; }