/* Set up the sync handle. Disables output_sync on error. */
static int
sync_init ()
{
int combined_output = 0;
#ifdef WINDOWS32
if ((!STREAM_OK (stdout) && !STREAM_OK (stderr))
|| (sync_handle = create_mutex ()) == -1)
{
perror_with_name ("output-sync suppressed: ", "stderr");
output_sync = 0;
}
else
{
combined_output = same_stream (stdout, stderr);
prepare_mutex_handle_string (sync_handle);
}
#else
if (STREAM_OK (stdout))
{
struct stat stbuf_o, stbuf_e;
sync_handle = fileno (stdout);
combined_output = (fstat (fileno (stdout), &stbuf_o) == 0
&& fstat (fileno (stderr), &stbuf_e) == 0
&& stbuf_o.st_dev == stbuf_e.st_dev
&& stbuf_o.st_ino == stbuf_e.st_ino);
}
else if (STREAM_OK (stderr))
sync_handle = fileno (stderr);
else
{
perror_with_name ("output-sync suppressed: ", "stderr");
output_sync = 0;
}
#endif
return combined_output;
}
static GstFlowReturn process_audio_packet(MpegTSDemuxer *demuxer, AVPacket *packet)
{
GstFlowReturn result = GST_FLOW_OK;
Stream *stream = &demuxer->audio;
if (!same_stream(demuxer, stream, packet))
return result;
GstBuffer *buffer = NULL;
GstEvent *newsegment_event = NULL;
void *buffer_data = av_mallocz(packet->size);
if (buffer_data != NULL)
{
memcpy(buffer_data, packet->data, packet->size);
buffer = gst_buffer_new_wrapped_full(0, buffer_data, packet->size, 0, packet->size, buffer_data, &av_free);
if (packet->pts != AV_NOPTS_VALUE)
{
if (demuxer->base_pts == GST_CLOCK_TIME_NONE)
{
demuxer->base_pts = PTS_TO_GSTTIME(packet->pts) + stream->offset_time;
}
gint64 time = PTS_TO_GSTTIME(packet->pts) + stream->offset_time - demuxer->base_pts;
if (time < 0)
time = 0;
if (stream->last_time > 0 && time < (gint64) (stream->last_time - PTS_TO_GSTTIME(G_MAXUINT32)))
{
stream->offset_time += PTS_TO_GSTTIME(MAX_PTS + 1); // Wraparound occured
time = PTS_TO_GSTTIME(packet->pts) + stream->offset_time;
#ifdef VERBOSE_DEBUG_AUDIO
g_print("[Audio wraparound] updating offset_time to %lld\n", stream->offset_time);
#endif
}
#ifdef VERBOSE_DEBUG_AUDIO
g_print("[Audio]: pts=%lld(%.4f) time=%lld (%.4f) offset_time=%lld last_time=%lld\n",
PTS_TO_GSTTIME(packet->pts), (double) PTS_TO_GSTTIME(packet->pts) / GST_SECOND,
time, (double) time / GST_SECOND, stream->offset_time, stream->last_time);
#endif
stream->last_time = time;
GST_BUFFER_TIMESTAMP(buffer) = time;
}
if (packet->duration != 0)
GST_BUFFER_DURATION(buffer) = PTS_TO_GSTTIME(packet->duration);
g_mutex_lock(&demuxer->lock);
stream->segment.position = GST_BUFFER_TIMESTAMP(buffer);
if (stream->discont)
{
GstSegment newsegment;
gst_segment_init(&newsegment, GST_FORMAT_TIME);
newsegment.flags = stream->segment.flags;
newsegment.rate = stream->segment.rate;
newsegment.start = stream->segment.time;
newsegment.stop = stream->segment.stop;
newsegment.time = stream->segment.time;
newsegment.position = stream->segment.position;
newsegment_event = gst_event_new_segment(&newsegment);
GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DISCONT);
stream->discont = FALSE;
#ifdef DEBUG_OUTPUT
g_print("MpegTS: [Audio] NEWSEGMENT: last_stop = %.4f\n", (double) stream->segment.last_stop / GST_SECOND);
#endif
}
g_mutex_unlock(&demuxer->lock);
} else
result = GST_FLOW_ERROR;
if (newsegment_event)
result = gst_pad_push_event(stream->sourcepad, newsegment_event) ? GST_FLOW_OK : GST_FLOW_FLUSHING;
if (result == GST_FLOW_OK)
result = gst_pad_push(stream->sourcepad, buffer);
else
gst_buffer_unref(buffer);
#ifdef VERBOSE_DEBUG_AUDIO
if (result != GST_FLOW_OK)
g_print("MpegTS: Audio push failed: %s\n", gst_flow_get_name(result));
#endif
return result;
}
