/**
* gst_base_rtp_audio_payload_flush:
* @baseaudiopayload: a #GstBaseRTPPayload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of the adapter as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
* -1, the timestamp will be calculated automatically.
*
* Returns: a #GstFlowReturn
*
* Since: 0.10.25
*/
GstFlowReturn
gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GstAdapter *adapter;
guint64 distance;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
if (payload_len == -1)
payload_len = gst_adapter_available (adapter);
/* nothing to do, just return */
if (payload_len == 0)
return GST_FLOW_OK;
if (timestamp == -1) {
/* calculate the timestamp */
timestamp = gst_adapter_prev_timestamp (adapter, &distance);
GST_LOG_OBJECT (baseaudiopayload,
"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
/* convert the number of bytes since the last timestamp to time and add to
* the last seen timestamp */
timestamp += priv->bytes_to_time (baseaudiopayload, distance);
}
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {
GstBuffer *buffer;
/* we can quickly take a buffer out of the adapter without having to copy
* anything. */
buffer = gst_adapter_take_buffer (adapter, payload_len);
ret = gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (adapter, payload, 0, payload_len);
gst_adapter_flush (adapter, payload_len);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}
static GstFlowReturn
gst_shape_wipe_video_sink_chain (GstPad * pad, GstBuffer * buffer)
{
GstShapeWipe *self = GST_SHAPE_WIPE (GST_PAD_PARENT (pad));
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *mask = NULL, *outbuf = NULL;
GstClockTime timestamp;
gboolean new_outbuf = FALSE;
if (G_UNLIKELY (self->fmt == GST_VIDEO_FORMAT_UNKNOWN))
return GST_FLOW_NOT_NEGOTIATED;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_stream_time (&self->segment, GST_FORMAT_TIME, timestamp);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
GST_DEBUG_OBJECT (self,
"Blending buffer with timestamp %" GST_TIME_FORMAT " at position %lf",
GST_TIME_ARGS (timestamp), self->mask_position);
g_mutex_lock (self->mask_mutex);
if (!self->mask)
g_cond_wait (self->mask_cond, self->mask_mutex);
if (self->mask == NULL) {
g_mutex_unlock (self->mask_mutex);
gst_buffer_unref (buffer);
return GST_FLOW_UNEXPECTED;
} else {
mask = gst_buffer_ref (self->mask);
}
g_mutex_unlock (self->mask_mutex);
if (!gst_shape_wipe_do_qos (self, GST_BUFFER_TIMESTAMP (buffer))) {
gst_buffer_unref (buffer);
gst_buffer_unref (mask);
return GST_FLOW_OK;
}
/* Try to blend inplace, if it's not possible
* get a new buffer from downstream.
*/
if (!gst_buffer_is_writable (buffer)) {
ret =
gst_pad_alloc_buffer_and_set_caps (self->srcpad, GST_BUFFER_OFFSET_NONE,
GST_BUFFER_SIZE (buffer), GST_PAD_CAPS (self->srcpad), &outbuf);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
gst_buffer_unref (buffer);
gst_buffer_unref (mask);
return ret;
}
gst_buffer_copy_metadata (outbuf, buffer, GST_BUFFER_COPY_ALL);
new_outbuf = TRUE;
} else {
outbuf = buffer;
}
if (self->fmt == GST_VIDEO_FORMAT_AYUV && self->mask_bpp == 16)
ret = gst_shape_wipe_blend_ayuv_16 (self, buffer, mask, outbuf);
else if (self->fmt == GST_VIDEO_FORMAT_AYUV)
ret = gst_shape_wipe_blend_ayuv_8 (self, buffer, mask, outbuf);
else if (self->fmt == GST_VIDEO_FORMAT_ARGB && self->mask_bpp == 16)
ret = gst_shape_wipe_blend_argb_16 (self, buffer, mask, outbuf);
else if (self->fmt == GST_VIDEO_FORMAT_ARGB)
ret = gst_shape_wipe_blend_argb_8 (self, buffer, mask, outbuf);
else if (self->fmt == GST_VIDEO_FORMAT_BGRA && self->mask_bpp == 16)
ret = gst_shape_wipe_blend_bgra_16 (self, buffer, mask, outbuf);
else if (self->fmt == GST_VIDEO_FORMAT_BGRA)
ret = gst_shape_wipe_blend_bgra_8 (self, buffer, mask, outbuf);
else
g_assert_not_reached ();
gst_buffer_unref (mask);
if (new_outbuf)
gst_buffer_unref (buffer);
if (ret != GST_FLOW_OK) {
gst_buffer_unref (outbuf);
return ret;
}
ret = gst_pad_push (self->srcpad, outbuf);
return ret;
}
static GstFlowReturn
gst_videoframe_audiolevel_asink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstClockTime timestamp, cur_time;
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GstBuffer *buf;
gsize inbuf_size;
guint64 start_offset, end_offset;
GstClockTime running_time;
gint rate, bpf;
gboolean discont = FALSE;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
running_time =
gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME, timestamp);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
bpf = GST_AUDIO_INFO_BPF (&self->ainfo);
start_offset = gst_util_uint64_scale (timestamp, rate, GST_SECOND);
inbuf_size = gst_buffer_get_size (inbuf);
end_offset = start_offset + inbuf_size / bpf;
g_mutex_lock (&self->mutex);
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| self->first_time == GST_CLOCK_TIME_NONE) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, rate, GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = timestamp;
} else if (timestamp - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (self->next_offset != -1)
GST_INFO_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
self->total_frames = 0;
self->first_time = running_time;
self->next_offset = end_offset;
} else {
self->next_offset += inbuf_size / bpf;
}
gst_adapter_push (self->adapter, gst_buffer_ref (inbuf));
GST_DEBUG_OBJECT (self, "Queue length %i",
g_queue_get_length (&self->vtimeq));
while (TRUE) {
GstClockTime *vt0, *vt1;
GstClockTime vtemp;
GstMessage *msg;
gsize bytes, available_bytes;
vtemp = GST_CLOCK_TIME_NONE;
while (!(g_queue_get_length (&self->vtimeq) >= 2 || self->video_eos_flag
|| self->audio_flush_flag || self->shutdown_flag))
g_cond_wait (&self->cond, &self->mutex);
if (self->audio_flush_flag || self->shutdown_flag) {
g_mutex_unlock (&self->mutex);
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
} else if (self->video_eos_flag) {
GST_DEBUG_OBJECT (self, "Video EOS flag alert");
/* nothing to do here if queue is empty */
if (g_queue_get_length (&self->vtimeq) == 0)
break;
if (g_queue_get_length (&self->vtimeq) < 2) {
vtemp = self->vsegment.position;
} else if (self->vsegment.position == GST_CLOCK_TIME_NONE) {
/* g_queue_get_length is surely >= 2 at this point
* so the adapter isn't empty */
buf =
gst_adapter_take_buffer (self->adapter,
gst_adapter_available (self->adapter));
if (buf != NULL) {
GstMessage *msg;
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
gst_buffer_unref (buf);
g_mutex_lock (&self->mutex); /* we unlock again later */
}
break;
}
} else if (g_queue_get_length (&self->vtimeq) < 2) {
continue;
}
vt0 = g_queue_pop_head (&self->vtimeq);
if (vtemp == GST_CLOCK_TIME_NONE)
vt1 = g_queue_peek_head (&self->vtimeq);
else
vt1 = &vtemp;
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
GST_DEBUG_OBJECT (self,
"Processing: current time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (cur_time));
GST_DEBUG_OBJECT (self, "Total frames is %i with a rate of %d",
self->total_frames, rate);
GST_DEBUG_OBJECT (self, "Start time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->first_time));
GST_DEBUG_OBJECT (self, "Time on top is %" GST_TIME_FORMAT,
GST_TIME_ARGS (*vt0));
if (cur_time < *vt0) {
guint num_frames =
gst_util_uint64_scale (*vt0 - cur_time, rate, GST_SECOND);
bytes = num_frames * GST_AUDIO_INFO_BPF (&self->ainfo);
available_bytes = gst_adapter_available (self->adapter);
if (available_bytes == 0) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
if (bytes == 0) {
cur_time = *vt0;
} else {
GST_DEBUG_OBJECT (self,
"Flushed %" G_GSIZE_FORMAT " out of %" G_GSIZE_FORMAT " bytes",
bytes, available_bytes);
gst_adapter_flush (self->adapter, MIN (bytes, available_bytes));
self->total_frames += num_frames;
if (available_bytes <= bytes) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
}
}
if (*vt1 > cur_time) {
bytes =
GST_AUDIO_INFO_BPF (&self->ainfo) * gst_util_uint64_scale (*vt1 -
cur_time, rate, GST_SECOND);
} else {
bytes = 0; /* We just need to discard vt0 */
}
available_bytes = gst_adapter_available (self->adapter);
GST_DEBUG_OBJECT (self,
"Adapter contains %" G_GSIZE_FORMAT " out of %" G_GSIZE_FORMAT " bytes",
available_bytes, bytes);
if (available_bytes < bytes) {
g_queue_push_head (&self->vtimeq, vt0);
goto done;
}
if (bytes > 0) {
buf = gst_adapter_take_buffer (self->adapter, bytes);
g_assert (buf != NULL);
} else {
/* Just an empty buffer */
buf = gst_buffer_new ();
}
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
g_mutex_lock (&self->mutex);
gst_buffer_unref (buf);
g_free (vt0);
if (available_bytes == bytes)
break;
}
done:
g_mutex_unlock (&self->mutex);
return gst_pad_push (self->asrcpad, inbuf);
}
/* Called when pipeline is pre-rolled */
static void
discoverer_collect (GstDiscoverer * dc)
{
GST_DEBUG ("Collecting information");
/* Stop the timeout handler if present */
if (dc->priv->timeoutid) {
g_source_remove (dc->priv->timeoutid);
dc->priv->timeoutid = 0;
}
if (dc->priv->streams) {
/* FIXME : Make this querying optional */
if (TRUE) {
GstElement *pipeline = (GstElement *) dc->priv->pipeline;
GstFormat format = GST_FORMAT_TIME;
gint64 dur;
GST_DEBUG ("Attempting to query duration");
if (gst_element_query_duration (pipeline, &format, &dur)) {
if (format == GST_FORMAT_TIME) {
GST_DEBUG ("Got duration %" GST_TIME_FORMAT, GST_TIME_ARGS (dur));
dc->priv->current_info->duration = (guint64) dur;
}
}
if (dc->priv->seeking_query) {
if (gst_element_query (pipeline, dc->priv->seeking_query)) {
gboolean seekable;
gst_query_parse_seeking (dc->priv->seeking_query, &format,
&seekable, NULL, NULL);
if (format == GST_FORMAT_TIME) {
GST_DEBUG ("Got seekable %d", seekable);
dc->priv->current_info->seekable = seekable;
}
}
}
}
if (dc->priv->current_topology)
dc->priv->current_info->stream_info = parse_stream_topology (dc,
dc->priv->current_topology, NULL);
/*
* Images need some special handling. They do not have a duration, have
* caps named image/<foo> (th exception being MJPEG video which is also
* type image/jpeg), and should consist of precisely one stream (actually
* initially there are 2, the image and raw stream, but we squash these
* while parsing the stream topology). At some ponit, if we find that these
* conditions are not sufficient, we can count the number of decoders and
* parsers in the chain, and if there's more than one decoder, or any
* parser at all, we should not mark this as an image.
*/
if (dc->priv->current_info->duration == 0 &&
dc->priv->current_info->stream_info != NULL &&
dc->priv->current_info->stream_info->next == NULL) {
GstStructure *st =
gst_caps_get_structure (dc->priv->current_info->stream_info->caps, 0);
if (g_str_has_prefix (gst_structure_get_name (st), "image/"))
((GstDiscovererVideoInfo *) dc->priv->current_info->
stream_info)->is_image = TRUE;
}
}
if (dc->priv->async) {
GST_DEBUG ("Emitting 'discoverered'");
g_signal_emit (dc, gst_discoverer_signals[SIGNAL_DISCOVERED], 0,
dc->priv->current_info, dc->priv->current_error);
/* Clients get a copy of current_info since it is a boxed type */
gst_discoverer_info_unref (dc->priv->current_info);
}
}
static gboolean
gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret = TRUE;
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* we need time for now */
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (jitterbuffer,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
/* now configure the values, we need these to time the release of the
* buffers on the srcpad. */
gst_segment_set_newsegment_full (&priv->segment, update,
rate, arate, format, start, stop, time);
/* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
case GST_EVENT_FLUSH_START:
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
ret = gst_pad_push_event (priv->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
ret = gst_pad_push_event (priv->srcpad, event);
ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
break;
case GST_EVENT_EOS:
{
/* push EOS in queue. We always push it at the head */
JBUF_LOCK (priv);
/* check for flushing, we need to discard the event and return FALSE when
* we are flushing */
ret = priv->srcresult == GST_FLOW_OK;
if (ret && !priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
priv->eos = TRUE;
JBUF_SIGNAL (priv);
} else if (priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
gst_flow_get_name (priv->srcresult));
}
JBUF_UNLOCK (priv);
gst_event_unref (event);
break;
}
default:
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
done:
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
ret = FALSE;
goto done;
}
}
GstIsoffParserResult
gst_isoff_sidx_parser_add_buffer (GstSidxParser * parser, GstBuffer * buffer,
guint * consumed)
{
GstIsoffParserResult res = GST_ISOFF_PARSER_OK;
GstByteReader reader;
GstMapInfo info;
gsize remaining;
guint32 fourcc;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
*consumed = 0;
return GST_ISOFF_PARSER_ERROR;
}
gst_byte_reader_init (&reader, info.data, info.size);
switch (parser->status) {
case GST_ISOFF_SIDX_PARSER_INIT:
if (gst_byte_reader_get_remaining (&reader) < GST_ISOFF_FULL_BOX_SIZE) {
break;
}
parser->size = gst_byte_reader_get_uint32_be_unchecked (&reader);
fourcc = gst_byte_reader_get_uint32_le_unchecked (&reader);
if (fourcc != GST_ISOFF_FOURCC_SIDX) {
res = GST_ISOFF_PARSER_UNEXPECTED;
gst_byte_reader_set_pos (&reader, 0);
break;
}
if (parser->size == 1) {
if (gst_byte_reader_get_remaining (&reader) < 12) {
gst_byte_reader_set_pos (&reader, 0);
break;
}
parser->size = gst_byte_reader_get_uint64_be_unchecked (&reader);
}
if (parser->size == 0) {
res = GST_ISOFF_PARSER_ERROR;
gst_byte_reader_set_pos (&reader, 0);
break;
}
parser->sidx.version = gst_byte_reader_get_uint8_unchecked (&reader);
parser->sidx.flags = gst_byte_reader_get_uint24_le_unchecked (&reader);
parser->status = GST_ISOFF_SIDX_PARSER_HEADER;
case GST_ISOFF_SIDX_PARSER_HEADER:
remaining = gst_byte_reader_get_remaining (&reader);
if (remaining < 12 + (parser->sidx.version == 0 ? 8 : 16)) {
break;
}
parser->sidx.ref_id = gst_byte_reader_get_uint32_be_unchecked (&reader);
parser->sidx.timescale =
gst_byte_reader_get_uint32_be_unchecked (&reader);
if (parser->sidx.version == 0) {
parser->sidx.earliest_pts =
gst_byte_reader_get_uint32_be_unchecked (&reader);
parser->sidx.first_offset =
gst_byte_reader_get_uint32_be_unchecked (&reader);
} else {
parser->sidx.earliest_pts =
gst_byte_reader_get_uint64_be_unchecked (&reader);
parser->sidx.first_offset =
gst_byte_reader_get_uint64_be_unchecked (&reader);
}
/* skip 2 reserved bytes */
gst_byte_reader_skip_unchecked (&reader, 2);
parser->sidx.entries_count =
gst_byte_reader_get_uint16_be_unchecked (&reader);
GST_LOG ("Timescale: %" G_GUINT32_FORMAT, parser->sidx.timescale);
GST_LOG ("Earliest pts: %" G_GUINT64_FORMAT, parser->sidx.earliest_pts);
GST_LOG ("First offset: %" G_GUINT64_FORMAT, parser->sidx.first_offset);
parser->cumulative_pts =
gst_util_uint64_scale_int_round (parser->sidx.earliest_pts,
GST_SECOND, parser->sidx.timescale);
if (parser->sidx.entries_count) {
parser->sidx.entries =
g_malloc (sizeof (GstSidxBoxEntry) * parser->sidx.entries_count);
}
parser->sidx.entry_index = 0;
parser->status = GST_ISOFF_SIDX_PARSER_DATA;
case GST_ISOFF_SIDX_PARSER_DATA:
while (parser->sidx.entry_index < parser->sidx.entries_count) {
GstSidxBoxEntry *entry =
&parser->sidx.entries[parser->sidx.entry_index];
remaining = gst_byte_reader_get_remaining (&reader);
if (remaining < 12)
break;
entry->offset = parser->cumulative_entry_size;
entry->pts = parser->cumulative_pts;
gst_isoff_parse_sidx_entry (entry, &reader);
entry->duration = gst_util_uint64_scale_int_round (entry->duration,
GST_SECOND, parser->sidx.timescale);
parser->cumulative_entry_size += entry->size;
parser->cumulative_pts += entry->duration;
GST_LOG ("Sidx entry %d) offset: %" G_GUINT64_FORMAT ", pts: %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT " - size %"
G_GUINT32_FORMAT, parser->sidx.entry_index, entry->offset,
GST_TIME_ARGS (entry->pts), GST_TIME_ARGS (entry->duration),
entry->size);
parser->sidx.entry_index++;
}
if (parser->sidx.entry_index == parser->sidx.entries_count)
parser->status = GST_ISOFF_SIDX_PARSER_FINISHED;
else
break;
case GST_ISOFF_SIDX_PARSER_FINISHED:
parser->sidx.entry_index = 0;
res = GST_ISOFF_PARSER_DONE;
break;
}
*consumed = gst_byte_reader_get_pos (&reader);
gst_buffer_unmap (buffer, &info);
return res;
}
/* Delayed seek callback. This gets called by the timer setup in the above function. */
static gboolean delayed_seek_cb (CustomData *data) {
GST_DEBUG ("Doing delayed seek to %" GST_TIME_FORMAT, GST_TIME_ARGS (data->desired_position));
execute_seek (data->desired_position, data);
return FALSE;
}
static gboolean
gst_base_video_decoder_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseVideoDecoder *base_video_decoder;
gboolean res = FALSE;
base_video_decoder = GST_BASE_VIDEO_DECODER (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
if (!base_video_decoder->packetized)
gst_base_video_decoder_drain (base_video_decoder, TRUE);
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SRC_PAD
(base_video_decoder), event);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
double rate;
double applied_rate;
GstFormat format;
gint64 start;
gint64 stop;
gint64 position;
GstSegment *segment = &base_video_decoder->segment;
gst_event_parse_new_segment_full (event, &update, &rate,
&applied_rate, &format, &start, &stop, &position);
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
if (!update) {
gst_base_video_decoder_flush (base_video_decoder);
}
base_video_decoder->timestamp_offset = start;
gst_segment_set_newsegment_full (segment,
update, rate, applied_rate, format, start, stop, position);
base_video_decoder->have_segment = TRUE;
GST_WARNING ("new segment: format %d rate %g start %" GST_TIME_FORMAT
" stop %" GST_TIME_FORMAT
" position %" GST_TIME_FORMAT
" update %d",
format, rate,
GST_TIME_ARGS (segment->start),
GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), update);
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SRC_PAD
(base_video_decoder), event);
break;
}
case GST_EVENT_FLUSH_STOP:
gst_base_video_decoder_flush (base_video_decoder);
gst_segment_init (&base_video_decoder->segment, GST_FORMAT_TIME);
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SRC_PAD
(base_video_decoder), event);
break;
default:
res = gst_pad_event_default (pad, event);
break;
}
done:
gst_object_unref (base_video_decoder);
return res;
newseg_wrong_format:
GST_DEBUG_OBJECT (base_video_decoder, "received non TIME newsegment");
gst_event_unref (event);
goto done;
}
static gboolean
gst_base_video_decoder_src_event (GstPad * pad, GstEvent * event)
{
GstBaseVideoDecoder *base_video_decoder;
gboolean res = FALSE;
base_video_decoder = GST_BASE_VIDEO_DECODER (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
/* FIXME: do seek using bitrate incase upstream doesn't handle it */
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SINK_PAD
(base_video_decoder), event);
break;
case GST_EVENT_QOS:
{
gdouble proportion;
GstClockTimeDiff diff;
GstClockTime timestamp;
GstClockTime duration;
gst_event_parse_qos (event, &proportion, &diff, ×tamp);
GST_OBJECT_LOCK (base_video_decoder);
base_video_decoder->proportion = proportion;
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp))) {
if (G_UNLIKELY (diff > 0)) {
if (base_video_decoder->state.fps_n > 0)
duration =
gst_util_uint64_scale (GST_SECOND,
base_video_decoder->state.fps_d,
base_video_decoder->state.fps_n);
else
duration = 0;
base_video_decoder->earliest_time = timestamp + 2 * diff + duration;
} else {
base_video_decoder->earliest_time = timestamp + diff;
}
} else {
base_video_decoder->earliest_time = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (base_video_decoder);
GST_DEBUG_OBJECT (base_video_decoder,
"got QoS %" GST_TIME_FORMAT ", %" G_GINT64_FORMAT ", %g",
GST_TIME_ARGS (timestamp), diff, proportion);
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SINK_PAD
(base_video_decoder), event);
break;
}
default:
res =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SINK_PAD
(base_video_decoder), event);
break;
}
gst_object_unref (base_video_decoder);
return res;
}
static GstFlowReturn
gst_mve_demux_chain (GstPad * sinkpad, GstBuffer * inbuf)
{
GstMveDemux *mve = GST_MVE_DEMUX (GST_PAD_PARENT (sinkpad));
GstFlowReturn ret = GST_FLOW_OK;
gst_adapter_push (mve->adapter, inbuf);
GST_DEBUG_OBJECT (mve, "queuing buffer, needed:%d, available:%u",
mve->needed_bytes, gst_adapter_available (mve->adapter));
while ((gst_adapter_available (mve->adapter) >= mve->needed_bytes) &&
(ret == GST_FLOW_OK)) {
GstMveDemuxStream *stream = NULL;
GstBuffer *outbuf = NULL;
switch (mve->state) {
case MVEDEMUX_STATE_INITIAL:
gst_adapter_flush (mve->adapter, mve->needed_bytes);
mve->chunk_offset += mve->needed_bytes;
mve->needed_bytes = 4;
mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
break;
case MVEDEMUX_STATE_NEXT_CHUNK:{
const guint8 *data;
guint16 size;
data = gst_adapter_peek (mve->adapter, mve->needed_bytes);
size = GST_MVE_SEGMENT_SIZE (data);
if (mve->chunk_offset >= mve->chunk_size) {
/* new chunk, flush buffer and proceed with next segment */
guint16 chunk_type = GST_READ_UINT16_LE (data + 2);
gst_adapter_flush (mve->adapter, mve->needed_bytes);
mve->chunk_size = size;
mve->chunk_offset = 0;
if (chunk_type > MVE_CHUNK_END) {
GST_WARNING_OBJECT (mve,
"skipping unknown chunk type 0x%02x of size:%u", chunk_type,
size);
mve->needed_bytes += size;
mve->state = MVEDEMUX_STATE_SKIP;
} else {
GST_DEBUG_OBJECT (mve, "found new chunk type 0x%02x of size:%u",
chunk_type, size);
}
} else if (mve->chunk_offset <= mve->chunk_size) {
/* new segment */
GST_DEBUG_OBJECT (mve, "found segment type 0x%02x of size:%u",
GST_MVE_SEGMENT_TYPE (data), size);
mve->needed_bytes += size;
mve->state = MVEDEMUX_STATE_MOVIE;
}
}
break;
case MVEDEMUX_STATE_MOVIE:
ret = gst_mve_parse_segment (mve, &stream, &outbuf);
if ((ret == GST_FLOW_OK) && (outbuf != NULL)) {
/* send buffer */
GST_DEBUG_OBJECT (mve,
"pushing buffer with time %" GST_TIME_FORMAT
" (%u bytes) on pad %s",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_SIZE (outbuf), GST_PAD_NAME (stream->pad));
ret = gst_pad_push (stream->pad, outbuf);
stream->last_flow = ret;
}
if (ret == GST_FLOW_NOT_LINKED) {
if (mve->audio_stream
&& mve->audio_stream->last_flow != GST_FLOW_NOT_LINKED)
ret = GST_FLOW_OK;
if (mve->video_stream
&& mve->video_stream->last_flow != GST_FLOW_NOT_LINKED)
ret = GST_FLOW_OK;
}
/* update current offset */
mve->chunk_offset += mve->needed_bytes;
mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
mve->needed_bytes = 4;
break;
case MVEDEMUX_STATE_SKIP:
mve->chunk_offset += mve->needed_bytes;
gst_adapter_flush (mve->adapter, mve->needed_bytes);
mve->state = MVEDEMUX_STATE_NEXT_CHUNK;
mve->needed_bytes = 4;
break;
default:
GST_ERROR_OBJECT (mve, "invalid state: %d", mve->state);
break;
}
}
return ret;
}
static gboolean
theora_parse_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstTheoraParse *parse;
gboolean res = FALSE;
parse = GST_THEORA_PARSE (parent);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
gint64 frame, value;
GstFormat my_format, format;
gint64 time;
frame = parse->prev_frame;
GST_LOG_OBJECT (parse,
"query %p: we have current frame: %" G_GINT64_FORMAT, query, frame);
/* parse format */
gst_query_parse_position (query, &format, NULL);
/* and convert to the final format in two steps with time as the
* intermediate step */
my_format = GST_FORMAT_TIME;
if (!(res =
theora_parse_src_convert (parse->sinkpad, GST_FORMAT_DEFAULT,
frame, &my_format, &time)))
goto error;
/* fixme: handle segments
time = (time - parse->segment.start) + parse->segment.time;
*/
GST_LOG_OBJECT (parse,
"query %p: our time: %" GST_TIME_FORMAT " (conv to %s)",
query, GST_TIME_ARGS (time), gst_format_get_name (format));
if (!(res =
theora_parse_src_convert (pad, my_format, time, &format, &value)))
goto error;
gst_query_set_position (query, format, value);
GST_LOG_OBJECT (parse,
"query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
format);
break;
}
case GST_QUERY_DURATION:
/* forward to peer for total */
if (!(res = gst_pad_query (GST_PAD_PEER (parse->sinkpad), query)))
goto error;
break;
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res =
theora_parse_src_convert (pad, src_fmt, src_val, &dest_fmt,
&dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
done:
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (parse, "query failed");
goto done;
}
}
gint
main (gint argc, gchar * argv[])
{
GstElement *pipeline, *filesrc, *decodebin;
GstStateChangeReturn res;
GstIterator *it;
GstBus *bus;
GValue data = { 0, };
gst_init (&argc, &argv);
pipeline = gst_pipeline_new ("pipeline");
filesrc = gst_element_factory_make ("filesrc", "filesrc");
g_assert (filesrc);
decodebin = gst_element_factory_make ("decodebin", "decodebin");
g_assert (decodebin);
gst_bin_add_many (GST_BIN (pipeline), filesrc, decodebin, NULL);
gst_element_link (filesrc, decodebin);
if (argc < 2) {
g_print ("usage: %s <filenames>\n", argv[0]);
exit (-1);
}
if (!g_str_has_prefix (argv[1], "file://")) {
g_object_set (G_OBJECT (filesrc), "location", argv[1], NULL);
} else {
g_object_set (G_OBJECT (filesrc), "location", argv[1] + 7, NULL);
}
/* we've got to connect fakesinks to newly decoded pads to make sure
* buffers have actually been flowing over those pads and caps have
* been set on them. decodebin might insert internal queues and
* without fakesinks it's pot-luck what caps we get from the pad, because
* it depends on whether the queues have started pushing buffers yet or not.
* With fakesinks we make sure that the pipeline doesn't go to PAUSED state
* before each fakesink has a buffer queued. */
g_signal_connect (decodebin, "new-decoded-pad",
G_CALLBACK (new_decoded_pad_cb), pipeline);
bus = gst_element_get_bus (pipeline);
g_print ("pause..\n");
res = gst_element_set_state (pipeline, GST_STATE_PAUSED);
if (res == GST_STATE_CHANGE_FAILURE) {
show_error ("Could not go to PAUSED state", bus);
exit (-1);
}
g_print ("waiting..\n");
res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
if (res != GST_STATE_CHANGE_SUCCESS) {
show_error ("Failed to complete state change to PAUSED", bus);
exit (-1);
}
g_print ("stats..\n");
it = gst_element_iterate_src_pads (decodebin);
while (gst_iterator_next (it, &data) == GST_ITERATOR_OK) {
GstPad *pad = g_value_get_object (&data);
GstCaps *caps;
gchar *str;
GstQuery *query;
g_print ("stream %s:\n", GST_OBJECT_NAME (pad));
caps = gst_pad_query_caps (pad, NULL);
str = gst_caps_to_string (caps);
g_print (" caps: %s\n", str);
g_free (str);
gst_caps_unref (caps);
query = gst_query_new_duration (GST_FORMAT_TIME);
if (gst_pad_query (pad, query)) {
gint64 duration;
gst_query_parse_duration (query, NULL, &duration);
g_print (" duration: %" GST_TIME_FORMAT "\n", GST_TIME_ARGS (duration));
}
gst_query_unref (query);
g_value_reset (&data);
}
g_value_unset (&data);
gst_iterator_free (it);
return 0;
}
static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstIdentity *identity = GST_IDENTITY (trans);
GstClockTime runtimestamp = G_GINT64_CONSTANT (0);
if (identity->check_perfect)
gst_identity_check_perfect (identity, buf);
if (identity->check_imperfect_timestamp)
gst_identity_check_imperfect_timestamp (identity, buf);
if (identity->check_imperfect_offset)
gst_identity_check_imperfect_offset (identity, buf);
/* update prev values */
identity->prev_timestamp = GST_BUFFER_TIMESTAMP (buf);
identity->prev_duration = GST_BUFFER_DURATION (buf);
identity->prev_offset_end = GST_BUFFER_OFFSET_END (buf);
identity->prev_offset = GST_BUFFER_OFFSET (buf);
if (identity->error_after >= 0) {
identity->error_after--;
if (identity->error_after == 0) {
GST_ELEMENT_ERROR (identity, CORE, FAILED,
(_("Failed after iterations as requested.")), (NULL));
return GST_FLOW_ERROR;
}
}
if (identity->drop_probability > 0.0) {
if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability) {
if (!identity->silent) {
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
identity->last_message =
g_strdup_printf
("dropping ******* (%s:%s)i (%d bytes, timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GINT64_FORMAT ", offset_end: % " G_GINT64_FORMAT
", flags: %d) %p", GST_DEBUG_PAD_NAME (trans->sinkpad),
GST_BUFFER_SIZE (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_OFFSET (buf),
GST_BUFFER_OFFSET_END (buf), GST_BUFFER_FLAGS (buf), buf);
GST_OBJECT_UNLOCK (identity);
gst_identity_notify_last_message (identity);
}
/* return DROPPED to basetransform. */
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
if (identity->dump) {
gst_util_dump_mem (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
if (!identity->silent) {
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
identity->last_message =
g_strdup_printf ("chain ******* (%s:%s)i (%d bytes, timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GINT64_FORMAT ", offset_end: % " G_GINT64_FORMAT ", flags: %d) %p",
GST_DEBUG_PAD_NAME (trans->sinkpad), GST_BUFFER_SIZE (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf),
GST_BUFFER_FLAGS (buf), buf);
GST_OBJECT_UNLOCK (identity);
gst_identity_notify_last_message (identity);
}
if (identity->datarate > 0) {
GstClockTime time = gst_util_uint64_scale_int (identity->offset,
GST_SECOND, identity->datarate);
GST_BUFFER_TIMESTAMP (buf) = time;
GST_BUFFER_DURATION (buf) =
GST_BUFFER_SIZE (buf) * GST_SECOND / identity->datarate;
}
if (identity->signal_handoffs)
g_signal_emit (identity, gst_identity_signals[SIGNAL_HANDOFF], 0, buf);
if (trans->segment.format == GST_FORMAT_TIME)
runtimestamp = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buf));
if ((identity->sync) && (trans->segment.format == GST_FORMAT_TIME)) {
GstClock *clock;
GST_OBJECT_LOCK (identity);
if ((clock = GST_ELEMENT (identity)->clock)) {
GstClockReturn cret;
GstClockTime timestamp;
timestamp = runtimestamp + GST_ELEMENT (identity)->base_time;
/* save id if we need to unlock */
/* FIXME: actually unlock this somewhere in the state changes */
identity->clock_id = gst_clock_new_single_shot_id (clock, timestamp);
GST_OBJECT_UNLOCK (identity);
cret = gst_clock_id_wait (identity->clock_id, NULL);
GST_OBJECT_LOCK (identity);
if (identity->clock_id) {
gst_clock_id_unref (identity->clock_id);
identity->clock_id = NULL;
}
if (cret == GST_CLOCK_UNSCHEDULED)
ret = GST_FLOW_UNEXPECTED;
}
GST_OBJECT_UNLOCK (identity);
}
identity->offset += GST_BUFFER_SIZE (buf);
if (identity->sleep_time && ret == GST_FLOW_OK)
g_usleep (identity->sleep_time);
if (identity->single_segment && (trans->segment.format == GST_FORMAT_TIME)
&& (ret == GST_FLOW_OK)) {
GST_BUFFER_TIMESTAMP (buf) = runtimestamp;
GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
}
return ret;
}
static GstFlowReturn
gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *payload;
GstBaseRTPAudioPayloadPrivate *priv;
guint payload_len;
GstFlowReturn ret;
guint available;
guint min_payload_len;
guint max_payload_len;
guint align;
guint size;
gboolean discont;
ret = GST_FLOW_OK;
payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
discont = GST_BUFFER_IS_DISCONT (buffer);
if (discont) {
GstClockTime timestamp;
GST_DEBUG_OBJECT (payload, "Got DISCONT");
/* flush everything out of the adapter, mark DISCONT */
ret = gst_base_rtp_audio_payload_flush (payload, -1, -1);
priv->discont = TRUE;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* get the distance between the timestamp gap and produce the same gap in
* the RTP timestamps */
if (priv->last_timestamp != -1 && timestamp != -1) {
/* we had a last timestamp, compare it to the new timestamp and update the
* offset counter for RTP timestamps. The effect is that we will produce
* output buffers containing the same RTP timestamp gap as the gap
* between the GST timestamps. */
if (timestamp > priv->last_timestamp) {
GstClockTime diff;
guint64 bytes;
/* we're only going to apply a positive gap, otherwise we let the marker
* bit do its thing. simply convert to bytes and add the the current
* offset */
diff = timestamp - priv->last_timestamp;
bytes = priv->time_to_bytes (payload, diff);
priv->offset += bytes;
GST_DEBUG_OBJECT (payload,
"elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT
", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,
priv->offset);
}
}
}
if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len,
&max_payload_len, &align))
goto config_error;
GST_DEBUG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
size = GST_BUFFER_SIZE (buffer);
/* shortcut, we don't need to use the adapter when the packet can be pushed
* through directly. */
available = gst_adapter_available (priv->adapter);
GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u",
size, available);
if (available == 0 && (size >= min_payload_len && size <= max_payload_len) &&
(size % align == 0)) {
/* If buffer fits on an RTP packet, let's just push it through
* this will check against max_ptime and max_mtu */
GST_DEBUG_OBJECT (payload, "Fast packet push");
ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer);
} else {
/* push the buffer in the adapter */
gst_adapter_push (priv->adapter, buffer);
available += size;
GST_DEBUG_OBJECT (payload, "available now %u", available);
/* as long as we have full frames */
while (available >= min_payload_len) {
/* get multiple of alignment */
payload_len = MIN (max_payload_len, available);
payload_len = ALIGN_DOWN (payload_len, align);
/* and flush out the bytes from the adapter, automatically set the
* timestamp. */
ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1);
available -= payload_len;
GST_DEBUG_OBJECT (payload, "available after push %u", available);
}
}
return ret;
/* ERRORS */
config_error:
{
GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not configure us properly"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
/* Reads in buffers, parses them, reframes into one-buffer-per-ogg-page, submits
* pages to output pad.
*/
static GstFlowReturn
gst_ogg_parse_chain (GstPad * pad, GstBuffer * buffer)
{
GstOggParse *ogg;
GstFlowReturn result = GST_FLOW_OK;
gint ret = -1;
guint32 serialno;
GstBuffer *pagebuffer;
GstClockTime buffertimestamp = GST_BUFFER_TIMESTAMP (buffer);
ogg = GST_OGG_PARSE (GST_OBJECT_PARENT (pad));
GST_LOG_OBJECT (ogg, "Chain function received buffer of size %d",
GST_BUFFER_SIZE (buffer));
gst_ogg_parse_submit_buffer (ogg, buffer);
while (ret != 0 && result == GST_FLOW_OK) {
ogg_page page;
/* We use ogg_sync_pageseek() rather than ogg_sync_pageout() so that we can
* track how many bytes the ogg layer discarded (in the case of sync errors,
* etc.); this allows us to accurately track the current stream offset
*/
ret = ogg_sync_pageseek (&ogg->sync, &page);
if (ret == 0) {
/* need more data, that's fine... */
break;
} else if (ret < 0) {
/* discontinuity; track how many bytes we skipped (-ret) */
ogg->offset -= ret;
} else {
gint64 granule = ogg_page_granulepos (&page);
#ifndef GST_DISABLE_GST_DEBUG
int bos = ogg_page_bos (&page);
#endif
guint64 startoffset = ogg->offset;
GstOggStream *stream;
gboolean keyframe;
serialno = ogg_page_serialno (&page);
stream = gst_ogg_parse_find_stream (ogg, serialno);
GST_LOG_OBJECT (ogg, "Timestamping outgoing buffer as %" GST_TIME_FORMAT,
GST_TIME_ARGS (buffertimestamp));
if (stream) {
buffertimestamp = gst_ogg_stream_get_end_time_for_granulepos (stream,
granule);
if (ogg->video_stream) {
if (stream == ogg->video_stream) {
keyframe = gst_ogg_stream_granulepos_is_key_frame (stream, granule);
} else {
keyframe = FALSE;
}
} else {
keyframe = TRUE;
}
} else {
buffertimestamp = GST_CLOCK_TIME_NONE;
keyframe = TRUE;
}
pagebuffer = gst_ogg_parse_buffer_from_page (&page, startoffset,
buffertimestamp);
/* We read out 'ret' bytes, so we set the next offset appropriately */
ogg->offset += ret;
GST_LOG_OBJECT (ogg,
"processing ogg page (serial %08x, pageno %ld, "
"granule pos %" G_GUINT64_FORMAT ", bos %d, offset %"
G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT ") keyframe=%d",
serialno, ogg_page_pageno (&page),
granule, bos, startoffset, ogg->offset, keyframe);
if (ogg_page_bos (&page)) {
/* If we've seen this serialno before, this is technically an error,
* we log this case but accept it - this one replaces the previous
* stream with this serialno. We can do this since we're streaming, and
* not supporting seeking...
*/
GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);
if (stream != NULL) {
GST_LOG_OBJECT (ogg, "Incorrect stream; repeats serial number %u "
"at offset %" G_GINT64_FORMAT, serialno, ogg->offset);
}
if (ogg->last_page_not_bos) {
GST_LOG_OBJECT (ogg, "Deleting all referenced streams, found a new "
"chain starting with serial %u", serialno);
gst_ogg_parse_delete_all_streams (ogg);
}
stream = gst_ogg_parse_new_stream (ogg, &page);
ogg->last_page_not_bos = FALSE;
gst_buffer_ref (pagebuffer);
stream->headers = g_list_append (stream->headers, pagebuffer);
if (!ogg->in_headers) {
GST_LOG_OBJECT (ogg,
"Found start of new chain at offset %" G_GUINT64_FORMAT,
startoffset);
ogg->in_headers = 1;
}
/* For now, we just keep the header buffer in the stream->headers list;
* it actually gets output once we've collected the entire set
*/
} else {
/* Non-BOS page. Either: we're outside headers, and this isn't a
* header (normal data), outside headers and this is (error!), inside
* headers, this is (append header), or inside headers and this isn't
* (we've found the end of headers; flush the lot!)
*
* Before that, we flag that the last page seen (this one) was not a
* BOS page; that way we know that when we next see a BOS page it's a
* new chain, and we can flush all existing streams.
*/
page_type type;
GstOggStream *stream = gst_ogg_parse_find_stream (ogg, serialno);
if (!stream) {
GST_LOG_OBJECT (ogg,
"Non-BOS page unexpectedly found at %" G_GINT64_FORMAT,
ogg->offset);
goto failure;
}
ogg->last_page_not_bos = TRUE;
type = gst_ogg_parse_is_header (ogg, stream, &page);
if (type == PAGE_PENDING && ogg->in_headers) {
gst_buffer_ref (pagebuffer);
stream->unknown_pages = g_list_append (stream->unknown_pages,
pagebuffer);
} else if (type == PAGE_HEADER) {
if (!ogg->in_headers) {
GST_LOG_OBJECT (ogg, "Header page unexpectedly found outside "
"headers at offset %" G_GINT64_FORMAT, ogg->offset);
goto failure;
} else {
/* Append the header to the buffer list, after any unknown previous
* pages
*/
stream->headers = g_list_concat (stream->headers,
stream->unknown_pages);
g_list_free (stream->unknown_pages);
gst_buffer_ref (pagebuffer);
stream->headers = g_list_append (stream->headers, pagebuffer);
}
} else { /* PAGE_DATA, or PAGE_PENDING but outside headers */
if (ogg->in_headers) {
/* First non-header page... set caps, flush headers.
*
* First up, we build a single GValue list of all the pagebuffers
* we're using for the headers, in order.
* Then we set this on the caps structure. Then we can start pushing
* buffers for the headers, and finally we send this non-header
* page.
*/
GstCaps *caps;
GstStructure *structure;
GValue array = { 0 };
gint count = 0;
gboolean found_pending_headers = FALSE;
GSList *l;
g_value_init (&array, GST_TYPE_ARRAY);
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
if (g_list_length (stream->headers) == 0) {
GST_LOG_OBJECT (ogg, "No primary header found for stream %08lx",
stream->serialno);
goto failure;
}
gst_ogg_parse_append_header (&array,
GST_BUFFER (stream->headers->data));
count++;
}
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *j;
/* already appended the first header, now do headers 2-N */
for (j = stream->headers->next; j != NULL; j = j->next) {
gst_ogg_parse_append_header (&array, GST_BUFFER (j->data));
count++;
}
}
caps = gst_pad_get_caps (ogg->srcpad);
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set_value (structure, "streamheader", &array);
gst_pad_set_caps (ogg->srcpad, caps);
g_value_unset (&array);
if (ogg->caps)
gst_caps_unref (ogg->caps);
ogg->caps = caps;
GST_LOG_OBJECT (ogg, "Set \"streamheader\" caps with %d buffers "
"(one per page)", count);
/* Now, we do the same thing, but push buffers... */
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GstBuffer *buf = GST_BUFFER (stream->headers->data);
buf = gst_buffer_make_metadata_writable (buf);
gst_buffer_set_caps (buf, caps);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *j;
/* pushed the first one for each stream already, now do 2-N */
for (j = stream->headers->next; j != NULL; j = j->next) {
GstBuffer *buf = GST_BUFFER (j->data);
buf = gst_buffer_make_metadata_writable (buf);
gst_buffer_set_caps (buf, caps);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
}
ogg->in_headers = 0;
/* And finally the pending data pages */
for (l = ogg->oggstreams; l != NULL; l = l->next) {
GstOggStream *stream = (GstOggStream *) l->data;
GList *k;
if (stream->unknown_pages == NULL)
continue;
if (found_pending_headers) {
GST_WARNING_OBJECT (ogg, "Incorrectly muxed headers found at "
"approximate offset %" G_GINT64_FORMAT, ogg->offset);
}
found_pending_headers = TRUE;
GST_LOG_OBJECT (ogg, "Pushing %d pending pages after headers",
g_list_length (stream->unknown_pages) + 1);
for (k = stream->unknown_pages; k != NULL; k = k->next) {
GstBuffer *buf;
buf = gst_buffer_make_metadata_writable (GST_BUFFER (k->data));
gst_buffer_set_caps (buf, caps);
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
}
g_list_foreach (stream->unknown_pages,
(GFunc) gst_mini_object_unref, NULL);
g_list_free (stream->unknown_pages);
stream->unknown_pages = NULL;
}
}
if (granule == -1) {
stream->stored_buffers = g_list_append (stream->stored_buffers,
pagebuffer);
} else {
while (stream->stored_buffers) {
GstBuffer *buf = stream->stored_buffers->data;
buf = gst_buffer_make_metadata_writable (buf);
gst_buffer_set_caps (buf, ogg->caps);
GST_BUFFER_TIMESTAMP (buf) = buffertimestamp;
if (!keyframe) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT);
} else {
keyframe = FALSE;
}
result = gst_pad_push (ogg->srcpad, buf);
if (result != GST_FLOW_OK)
return result;
stream->stored_buffers =
g_list_delete_link (stream->stored_buffers,
stream->stored_buffers);
}
pagebuffer = gst_buffer_make_metadata_writable (pagebuffer);
gst_buffer_set_caps (pagebuffer, ogg->caps);
if (!keyframe) {
GST_BUFFER_FLAG_SET (pagebuffer, GST_BUFFER_FLAG_DELTA_UNIT);
} else {
keyframe = FALSE;
}
result = gst_pad_push (ogg->srcpad, pagebuffer);
if (result != GST_FLOW_OK)
return result;
}
}
}
}
}
return result;
failure:
gst_pad_push_event (GST_PAD (ogg->srcpad), gst_event_new_eos ());
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_base_video_decoder_chain (GstPad * pad, GstBuffer * buf)
{
GstBaseVideoDecoder *base_video_decoder;
GstFlowReturn ret;
GST_DEBUG ("chain %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
#if 0
/* requiring the pad to be negotiated makes it impossible to use
* oggdemux or filesrc ! decoder */
if (!gst_pad_is_negotiated (pad)) {
GST_DEBUG ("not negotiated");
return GST_FLOW_NOT_NEGOTIATED;
}
#endif
base_video_decoder = GST_BASE_VIDEO_DECODER (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (base_video_decoder, "chain");
if (!base_video_decoder->have_segment) {
GstEvent *event;
GstFlowReturn ret;
GST_WARNING
("Received buffer without a new-segment. Assuming timestamps start from 0.");
gst_segment_set_newsegment_full (&base_video_decoder->segment,
FALSE, 1.0, 1.0, GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0);
base_video_decoder->have_segment = TRUE;
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0,
GST_CLOCK_TIME_NONE, 0);
ret =
gst_pad_push_event (GST_BASE_VIDEO_DECODER_SRC_PAD (base_video_decoder),
event);
if (!ret) {
GST_ERROR ("new segment event ret=%d", ret);
return GST_FLOW_ERROR;
}
}
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (base_video_decoder, "received DISCONT buffer");
gst_base_video_decoder_flush (base_video_decoder);
}
base_video_decoder->input_offset += GST_BUFFER_SIZE (buf);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
gst_base_video_decoder_add_timestamp (base_video_decoder, buf);
}
if (!base_video_decoder->current_frame)
base_video_decoder->current_frame =
gst_base_video_decoder_new_frame (base_video_decoder);
if (base_video_decoder->packetized) {
base_video_decoder->current_frame->sink_buffer = buf;
ret = gst_base_video_decoder_have_frame (base_video_decoder, TRUE, NULL);
} else {
gst_adapter_push (base_video_decoder->input_adapter, buf);
ret = gst_base_video_decoder_drain (base_video_decoder, FALSE);
}
gst_object_unref (base_video_decoder);
return ret;
}
static GstBuffer *
gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpDTMFDepay *rtpdtmfdepay = NULL;
GstBuffer *outbuf = NULL;
gint payload_len;
guint8 *payload = NULL;
guint32 timestamp;
GstRTPDTMFPayload dtmf_payload;
gboolean marker;
GstStructure *structure = NULL;
GstMessage *dtmf_message = NULL;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
if (payload_len != sizeof (GstRTPDTMFPayload))
goto bad_packet;
memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
if (dtmf_payload.event > MAX_EVENT)
goto bad_packet;
marker = gst_rtp_buffer_get_marker (buf);
timestamp = gst_rtp_buffer_get_timestamp (buf);
dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
/* clip to whole units of unit_time */
if (rtpdtmfdepay->unit_time) {
guint unit_time_clock =
(rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
if (dtmf_payload.duration % unit_time_clock) {
/* Make sure we don't overflow the duration */
if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
dtmf_payload.duration += unit_time_clock -
(dtmf_payload.duration % unit_time_clock);
else
dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
}
}
/* clip to max duration */
if (rtpdtmfdepay->max_duration) {
guint max_duration_clock =
(rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
if (max_duration_clock < G_MAXUINT16 &&
dtmf_payload.duration > max_duration_clock)
dtmf_payload.duration = max_duration_clock;
}
GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
"marker=%d - timestamp=%u - event=%d - duration=%d",
marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
GST_DEBUG_OBJECT (depayload,
"Previous information : timestamp=%u - duration=%d",
rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
/* First packet */
if (marker || rtpdtmfdepay->previous_ts != timestamp) {
rtpdtmfdepay->sample = 0;
rtpdtmfdepay->previous_ts = timestamp;
rtpdtmfdepay->previous_duration = dtmf_payload.duration;
rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);
structure = gst_structure_new ("dtmf-event",
"number", G_TYPE_INT, dtmf_payload.event,
"volume", G_TYPE_INT, dtmf_payload.volume,
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
if (structure) {
dtmf_message =
gst_message_new_element (GST_OBJECT (depayload), structure);
if (dtmf_message) {
if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
GST_ERROR_OBJECT (depayload,
"Unable to send dtmf-event message to bus");
}
} else {
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
}
} else {
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
}
} else {
guint16 duration = dtmf_payload.duration;
dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
/* If late buffer, ignore */
if (duration > rtpdtmfdepay->previous_duration)
rtpdtmfdepay->previous_duration = duration;
}
GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
" - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
rtpdtmfdepay->previous_duration, dtmf_payload.duration,
(rtpdtmfdepay->previous_duration - dtmf_payload.duration),
depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
/* If late or duplicate packet (like the redundant end packet). Ignore */
if (dtmf_payload.duration > 0) {
outbuf = gst_buffer_new ();
gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload, outbuf);
GST_BUFFER_TIMESTAMP (outbuf) = rtpdtmfdepay->first_gst_ts +
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
GST_SECOND / depayload->clock_rate;
GST_BUFFER_OFFSET (outbuf) =
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
GST_SECOND / depayload->clock_rate;
GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
GST_SECOND / depayload->clock_rate;
GST_DEBUG_OBJECT (depayload,
"timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
return outbuf;
bad_packet:
GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
("Packet did not validate"), (NULL));
return NULL;
}
GstFlowReturn
gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
GstVideoFrame * frame)
{
GstBaseVideoDecoderClass *base_video_decoder_class;
GstClockTime presentation_timestamp;
GstClockTime presentation_duration;
GstBuffer *src_buffer;
GST_DEBUG ("finish frame");
base_video_decoder_class =
GST_BASE_VIDEO_DECODER_GET_CLASS (base_video_decoder);
if (!gst_base_video_decoder_set_src_caps (base_video_decoder))
return GST_FLOW_NOT_NEGOTIATED;
gst_base_video_decoder_calculate_timestamps (base_video_decoder, frame,
&presentation_timestamp, &presentation_duration);
src_buffer = frame->src_buffer;
GST_BUFFER_FLAG_UNSET (src_buffer, GST_BUFFER_FLAG_DELTA_UNIT);
if (base_video_decoder->state.interlaced) {
#ifndef GST_VIDEO_BUFFER_TFF
#define GST_VIDEO_BUFFER_TFF (GST_MINI_OBJECT_FLAG_LAST << 5)
#endif
#ifndef GST_VIDEO_BUFFER_RFF
#define GST_VIDEO_BUFFER_RFF (GST_MINI_OBJECT_FLAG_LAST << 6)
#endif
#ifndef GST_VIDEO_BUFFER_ONEFIELD
#define GST_VIDEO_BUFFER_ONEFIELD (GST_MINI_OBJECT_FLAG_LAST << 7)
#endif
if (GST_VIDEO_FRAME_FLAG_IS_SET (frame, GST_VIDEO_FRAME_FLAG_TFF)) {
GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_TFF);
} else {
GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_TFF);
}
GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_RFF);
GST_BUFFER_FLAG_UNSET (src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
if (frame->n_fields == 3) {
GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_RFF);
} else if (frame->n_fields == 1) {
GST_BUFFER_FLAG_SET (src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
}
}
if (base_video_decoder->discont) {
GST_BUFFER_FLAG_UNSET (src_buffer, GST_BUFFER_FLAG_DISCONT);
base_video_decoder->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (src_buffer) = presentation_timestamp;
GST_BUFFER_DURATION (src_buffer) = presentation_duration;
GST_BUFFER_OFFSET (src_buffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (src_buffer) = GST_BUFFER_OFFSET_NONE;
GST_DEBUG ("pushing frame %" GST_TIME_FORMAT,
GST_TIME_ARGS (presentation_timestamp));
if (base_video_decoder->sink_clipping) {
gint64 start = GST_BUFFER_TIMESTAMP (src_buffer);
gint64 stop = GST_BUFFER_TIMESTAMP (src_buffer) +
GST_BUFFER_DURATION (src_buffer);
if (gst_segment_clip (&base_video_decoder->segment, GST_FORMAT_TIME,
start, stop, &start, &stop)) {
GST_BUFFER_TIMESTAMP (src_buffer) = start;
GST_BUFFER_DURATION (src_buffer) = stop - start;
GST_DEBUG ("accepting buffer inside segment: %" GST_TIME_FORMAT
" %" GST_TIME_FORMAT
" seg %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
" time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer) +
GST_BUFFER_DURATION (src_buffer)),
GST_TIME_ARGS (base_video_decoder->segment.start),
GST_TIME_ARGS (base_video_decoder->segment.stop),
GST_TIME_ARGS (base_video_decoder->segment.time));
} else {
GST_DEBUG ("dropping buffer outside segment: %" GST_TIME_FORMAT
" %" GST_TIME_FORMAT
" seg %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT
" time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (src_buffer) +
GST_BUFFER_DURATION (src_buffer)),
GST_TIME_ARGS (base_video_decoder->segment.start),
GST_TIME_ARGS (base_video_decoder->segment.stop),
GST_TIME_ARGS (base_video_decoder->segment.time));
gst_video_frame_unref (frame);
return GST_FLOW_OK;
}
}
gst_buffer_ref (src_buffer);
gst_video_frame_unref (frame);
if (base_video_decoder_class->shape_output)
return base_video_decoder_class->shape_output (base_video_decoder,
src_buffer);
return gst_pad_push (GST_BASE_VIDEO_DECODER_SRC_PAD (base_video_decoder),
src_buffer);
}
HRESULT AudioFakeSink::DoRenderSample(IMediaSample *pMediaSample)
{
GstBuffer *out_buf = NULL;
gboolean in_seg = FALSE;
GstClockTime buf_start, buf_stop;
gint64 clip_start = 0, clip_stop = 0;
guint start_offset = 0, stop_offset;
GstClockTime duration;
if(pMediaSample)
{
BYTE *pBuffer = NULL;
LONGLONG lStart = 0, lStop = 0;
long size = pMediaSample->GetActualDataLength();
pMediaSample->GetPointer(&pBuffer);
pMediaSample->GetTime(&lStart, &lStop);
if (!GST_CLOCK_TIME_IS_VALID (mDec->timestamp)) {
// Convert REFERENCE_TIME to GST_CLOCK_TIME
mDec->timestamp = (GstClockTime)lStart * 100;
}
duration = (lStop - lStart) * 100;
buf_start = mDec->timestamp;
buf_stop = mDec->timestamp + duration;
/* save stop position to start next buffer with it */
mDec->timestamp = buf_stop;
/* check if this buffer is in our current segment */
in_seg = gst_segment_clip (mDec->segment, GST_FORMAT_TIME,
buf_start, buf_stop, &clip_start, &clip_stop);
/* if the buffer is out of segment do not push it downstream */
if (!in_seg) {
GST_DEBUG_OBJECT (mDec,
"buffer is out of segment, start %" GST_TIME_FORMAT " stop %"
GST_TIME_FORMAT, GST_TIME_ARGS (buf_start), GST_TIME_ARGS (buf_stop));
goto done;
}
/* buffer is entirely or partially in-segment, so allocate a
* GstBuffer for output, and clip if required */
/* allocate a new buffer for raw audio */
mDec->last_ret = gst_pad_alloc_buffer (mDec->srcpad,
GST_BUFFER_OFFSET_NONE,
size,
GST_PAD_CAPS (mDec->srcpad), &out_buf);
if (!out_buf) {
GST_WARNING_OBJECT (mDec, "cannot allocate a new GstBuffer");
goto done;
}
/* set buffer properties */
GST_BUFFER_TIMESTAMP (out_buf) = buf_start;
GST_BUFFER_DURATION (out_buf) = duration;
memcpy (GST_BUFFER_DATA (out_buf), pBuffer,
MIN ((unsigned int)size, GST_BUFFER_SIZE (out_buf)));
/* we have to remove some heading samples */
if ((GstClockTime) clip_start > buf_start) {
start_offset = (guint)gst_util_uint64_scale_int (clip_start - buf_start,
mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
}
else
start_offset = 0;
/* we have to remove some trailing samples */
if ((GstClockTime) clip_stop < buf_stop) {
stop_offset = (guint)gst_util_uint64_scale_int (buf_stop - clip_stop,
mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
}
else
stop_offset = size;
/* truncating */
if ((start_offset != 0) || (stop_offset != (size_t) size)) {
GstBuffer *subbuf = gst_buffer_create_sub (out_buf, start_offset,
stop_offset - start_offset);
if (subbuf) {
gst_buffer_set_caps (subbuf, GST_PAD_CAPS (mDec->srcpad));
gst_buffer_unref (out_buf);
out_buf = subbuf;
}
}
GST_BUFFER_TIMESTAMP (out_buf) = clip_start;
GST_BUFFER_DURATION (out_buf) = clip_stop - clip_start;
/* replace the saved stop position by the clipped one */
mDec->timestamp = clip_stop;
GST_DEBUG_OBJECT (mDec,
"push_buffer (size %d)=> pts %" GST_TIME_FORMAT " stop %" GST_TIME_FORMAT
" duration %" GST_TIME_FORMAT, size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf) +
GST_BUFFER_DURATION (out_buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (out_buf)));
mDec->last_ret = gst_pad_push (mDec->srcpad, out_buf);
}
done:
return S_OK;
}
static gboolean
gst_base_video_parse_src_query (GstPad * pad, GstQuery * query)
{
GstBaseVideoParse *base_parse;
gboolean res = FALSE;
base_parse = GST_BASE_VIDEO_PARSE (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gint64 time;
gint64 value;
gst_query_parse_position (query, &format, NULL);
time = gst_util_uint64_scale (base_parse->presentation_frame_number,
base_parse->state.fps_n, base_parse->state.fps_d);
time += base_parse->segment.time;
GST_DEBUG ("query position %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
res = gst_base_video_encoded_video_convert (&base_parse->state,
GST_FORMAT_TIME, time, &format, &value);
if (!res)
goto error;
gst_query_set_position (query, format, value);
break;
}
case GST_QUERY_DURATION:
res =
gst_pad_query (GST_PAD_PEER (GST_BASE_VIDEO_CODEC_SINK_PAD
(base_parse)), query);
if (!res)
goto error;
break;
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_WARNING ("query convert");
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
res = gst_base_video_encoded_video_convert (&base_parse->state,
src_fmt, src_val, &dest_fmt, &dest_val);
if (!res)
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
done:
gst_object_unref (base_parse);
return res;
error:
GST_DEBUG_OBJECT (base_parse, "query failed");
goto done;
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime timestamp;
guint16 seqnum;
guint32 rtptime;
gboolean reset_seq, discont;
gint gap;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
priv = filter->priv;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
/* we must validate, it's possible that this element is plugged right after a
* network receiver and we don't want to operate on invalid data */
if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
goto invalid_buffer;
priv->discont = GST_BUFFER_IS_DISCONT (in);
timestamp = GST_BUFFER_TIMESTAMP (in);
/* convert to running_time and save the timestamp, this is the timestamp
* we put on outgoing buffers. */
timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
timestamp);
priv->timestamp = timestamp;
priv->duration = GST_BUFFER_DURATION (in);
seqnum = gst_rtp_buffer_get_seq (in);
rtptime = gst_rtp_buffer_get_timestamp (in);
reset_seq = TRUE;
discont = FALSE;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
GST_TIME_ARGS (timestamp));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are striclty increasing, dropping anything that is out of the ordinary. We
* can only do this when the next_seqnum is known. */
if (G_LIKELY (priv->next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
/* if we have no gap, all is fine */
if (G_UNLIKELY (gap != 0)) {
GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
priv->next_seqnum, gap);
if (gap < 0) {
/* seqnum > next_seqnum, we are missing some packets, this is always a
* DISCONT. */
GST_LOG_OBJECT (filter, "%d missing packets", gap);
discont = TRUE;
} else {
/* seqnum < next_seqnum, we have seen this packet before or the sender
* could be restarted. If the packet is not too old, we throw it away as
* a duplicate, otherwise we mark discont and continue. 100 misordered
* packets is a good threshold. See also RFC 4737. */
if (gap < 100)
goto dropping;
GST_LOG_OBJECT (filter,
"%d > 100, packet too old, sender likely restarted", gap);
discont = TRUE;
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
if (G_UNLIKELY (discont && !priv->discont)) {
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
/* we detected a seqnum discont but the buffer was not flagged with a discont,
* set the discont flag so that the subclass can throw away old data. */
priv->discont = TRUE;
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (G_UNLIKELY (bclass->process == NULL))
goto no_process;
/* let's send it out to processing */
out_buf = bclass->process (filter, in);
if (out_buf) {
/* we pass rtptime as backward compatibility, in reality, the incomming
* buffer timestamp is always applied to the outgoing packet. */
ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
}
gst_buffer_unref (in);
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, (NULL),
("Not RTP format was negotiated"));
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (in);
return GST_FLOW_OK;
}
dropping:
{
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_base_video_parse_src_event (GstPad * pad, GstEvent * event)
{
GstBaseVideoParse *base_video_parse;
gboolean res = FALSE;
base_video_parse = GST_BASE_VIDEO_PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
GstFormat format, tformat;
gdouble rate;
GstEvent *real_seek;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 tcur, tstop;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type,
&cur, &stop_type, &stop);
gst_event_unref (event);
tformat = GST_FORMAT_TIME;
res = gst_base_video_encoded_video_convert (&base_video_parse->state,
format, cur, &tformat, &tcur);
if (!res)
goto convert_error;
res = gst_base_video_encoded_video_convert (&base_video_parse->state,
format, stop, &tformat, &tstop);
if (!res)
goto convert_error;
real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
flags, cur_type, tcur, stop_type, tstop);
res =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SINK_PAD (base_video_parse),
real_seek);
break;
}
#if 0
case GST_EVENT_QOS:
{
gdouble proportion;
GstClockTimeDiff diff;
GstClockTime timestamp;
gst_event_parse_qos (event, &proportion, &diff, ×tamp);
GST_OBJECT_LOCK (base_video_parse);
base_video_parse->proportion = proportion;
base_video_parse->earliest_time = timestamp + diff;
GST_OBJECT_UNLOCK (base_video_parse);
GST_DEBUG_OBJECT (base_video_parse,
"got QoS %" GST_TIME_FORMAT ", %" G_GINT64_FORMAT,
GST_TIME_ARGS (timestamp), diff);
res = gst_pad_push_event (base_video_parse->sinkpad, event);
break;
}
#endif
default:
res =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SINK_PAD (base_video_parse),
event);
break;
}
done:
gst_object_unref (base_video_parse);
return res;
convert_error:
GST_DEBUG_OBJECT (base_video_parse, "could not convert format");
goto done;
}
/**
* This funcion will push out buffers on the source pad.
*
* For each pushed buffer, the seqnum is recorded, if the next buffer B has a
* different seqnum (missing packets before B), this function will wait for the
* missing packet to arrive up to the timestamp of buffer B.
*/
static void
gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn result;
guint16 seqnum;
guint32 next_seqnum;
GstClockTime timestamp, out_time;
gboolean discont = FALSE;
gint gap;
priv = jitterbuffer->priv;
JBUF_LOCK_CHECK (priv, flushing);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
while (TRUE) {
/* always wait if we are blocked */
if (!priv->blocked) {
/* if we have a packet, we can exit the loop and grab it */
if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (priv->eos)
goto do_eos;
}
/* underrun, wait for packets or flushing now */
priv->waiting = TRUE;
JBUF_WAIT_CHECK (priv, flushing);
priv->waiting = FALSE;
}
/* peek a buffer, we're just looking at the timestamp and the sequence number.
* If all is fine, we'll pop and push it. If the sequence number is wrong we
* wait on the timestamp. In the chain function we will unlock the wait when a
* new buffer is available. The peeked buffer is valid for as long as we hold
* the jitterbuffer lock. */
outbuf = rtp_jitter_buffer_peek (priv->jbuf);
/* get the seqnum and the next expected seqnum */
seqnum = gst_rtp_buffer_get_seq (outbuf);
next_seqnum = priv->next_seqnum;
/* get the timestamp, this is already corrected for clock skew by the
* jitterbuffer */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
GST_DEBUG_OBJECT (jitterbuffer,
"Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
rtp_jitter_buffer_num_packets (priv->jbuf));
/* apply our timestamp offset to the incomming buffer, this will be our output
* timestamp. */
out_time = apply_offset (jitterbuffer, timestamp);
/* get the gap between this and the previous packet. If we don't know the
* previous packet seqnum assume no gap. */
if (next_seqnum != -1) {
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
/* if we have a packet that we already pushed or considered dropped, pop it
* off and get the next packet */
if (gap < 0) {
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
seqnum, next_seqnum);
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (outbuf);
goto again;
}
} else {
GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
gap = -1;
}
/* If we don't know what the next seqnum should be (== -1) we have to wait
* because it might be possible that we are not receiving this buffer in-order,
* a buffer with a lower seqnum could arrive later and we want to push that
* earlier buffer before this buffer then.
* If we know the expected seqnum, we can compare it to the current seqnum to
* determine if we have missing a packet. If we have a missing packet (which
* must be before this packet) we can wait for it until the deadline for this
* packet expires. */
if (gap != 0 && out_time != -1) {
GstClockID id;
GstClockTime sync_time;
GstClockReturn ret;
GstClock *clock;
GstClockTime duration = GST_CLOCK_TIME_NONE;
if (gap > 0) {
/* we have a gap */
GST_WARNING_OBJECT (jitterbuffer,
"Sequence number GAP detected: expected %d instead of %d (%d missing)",
next_seqnum, seqnum, gap);
if (priv->last_out_time != -1) {
GST_DEBUG_OBJECT (jitterbuffer,
"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
/* interpolate between the current time and the last time based on
* number of packets we are missing, this is the estimated duration
* for the missing packet based on equidistant packet spacing. Also make
* sure we never go negative. */
if (out_time > priv->last_out_time)
duration = (out_time - priv->last_out_time) / (gap + 1);
else
goto lost;
GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
/* add this duration to the timestamp of the last packet we pushed */
out_time = (priv->last_out_time + duration);
}
} else {
/* we don't know what the next_seqnum should be, wait for the last
* possible moment to push this buffer, maybe we get an earlier seqnum
* while we wait */
GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
}
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
GST_OBJECT_UNLOCK (jitterbuffer);
/* let's just push if there is no clock */
goto push_buffer;
}
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time));
/* prepare for sync against clock */
sync_time = out_time + GST_ELEMENT_CAST (jitterbuffer)->base_time;
/* add latency, this includes our own latency and the peer latency. */
sync_time += (priv->latency_ms * GST_MSECOND);
sync_time += priv->peer_latency;
/* create an entry for the clock */
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
GST_OBJECT_UNLOCK (jitterbuffer);
/* release the lock so that the other end can push stuff or unlock */
JBUF_UNLOCK (priv);
ret = gst_clock_id_wait (id, NULL);
JBUF_LOCK (priv);
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
/* at this point, the clock could have been unlocked by a timeout, a new
* tail element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if (priv->srcresult != GST_FLOW_OK)
goto flushing;
/* if we got unscheduled and we are not flushing, it's because a new tail
* element became available in the queue. Grab it and try to push or sync. */
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
goto again;
}
lost:
/* we now timed out, this means we lost a packet or finished synchronizing
* on the first buffer. */
if (gap > 0) {
GstEvent *event;
/* we had a gap and thus we lost a packet. Create an event for this. */
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
priv->num_late++;
discont = TRUE;
if (priv->do_lost) {
/* create paket lost event */
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new ("GstRTPPacketLost",
"seqnum", G_TYPE_UINT, (guint) next_seqnum,
"timestamp", G_TYPE_UINT64, out_time,
"duration", G_TYPE_UINT64, duration, NULL));
gst_pad_push_event (priv->srcpad, event);
}
/* update our expected next packet */
priv->last_popped_seqnum = next_seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (next_seqnum + 1) & 0xffff;
/* look for next packet */
goto again;
}
/* there was no known gap,just the first packet, exit the loop and push */
GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
/* get new timestamp, latency might have changed */
out_time = apply_offset (jitterbuffer, timestamp);
}
push_buffer:
/* when we get here we are ready to pop and push the buffer */
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
if (discont || priv->discont) {
/* set DISCONT flag when we missed a packet. */
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* apply timestamp with offset to buffer now */
GST_BUFFER_TIMESTAMP (outbuf) = out_time;
/* now we are ready to push the buffer. Save the seqnum and release the lock
* so the other end can push stuff in the queue again. */
priv->last_popped_seqnum = seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (seqnum + 1) & 0xffff;
JBUF_UNLOCK (priv);
/* push buffer */
GST_DEBUG_OBJECT (jitterbuffer,
"Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (out_time));
result = gst_pad_push (priv->srcpad, outbuf);
if (result != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
do_eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
priv->srcresult = GST_FLOW_UNEXPECTED;
gst_pad_pause_task (priv->srcpad);
gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
JBUF_UNLOCK (priv);
return;
}
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
pause:
{
const gchar *reason = gst_flow_get_name (result);
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
JBUF_LOCK (priv);
/* store result */
priv->srcresult = result;
/* we don't post errors or anything because upstream will do that for us
* when we pass the return value upstream. */
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
}
static gboolean
gst_base_video_parse_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseVideoParse *base_video_parse;
gboolean ret = FALSE;
base_video_parse = GST_BASE_VIDEO_PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
ret =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_parse),
event);
break;
case GST_EVENT_FLUSH_STOP:
gst_base_video_parse_reset (base_video_parse);
ret =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_parse),
event);
break;
case GST_EVENT_EOS:
if (gst_base_video_parse_push_all (base_video_parse,
FALSE) == GST_FLOW_ERROR) {
gst_event_unref (event);
return FALSE;
}
ret =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_parse),
event);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
GstFormat format;
gdouble rate;
gint64 start, stop, time;
gst_event_parse_new_segment (event, &update, &rate, &format, &start,
&stop, &time);
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
if (rate <= 0.0)
goto newseg_wrong_rate;
GST_DEBUG ("newsegment %" GST_TIME_FORMAT " %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (time));
gst_segment_set_newsegment (&base_video_parse->segment, update,
rate, format, start, stop, time);
ret =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_parse),
event);
break;
}
default:
ret =
gst_pad_push_event (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_parse),
event);
break;
}
done:
gst_object_unref (base_video_parse);
return ret;
newseg_wrong_format:
GST_DEBUG_OBJECT (base_video_parse, "received non TIME newsegment");
gst_event_unref (event);
goto done;
newseg_wrong_rate:
GST_DEBUG_OBJECT (base_video_parse, "negative rates not supported");
gst_event_unref (event);
goto done;
}
static GstFlowReturn
gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
guint16 seqnum;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
guint64 latency_ts;
gboolean tail;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (!gst_rtp_buffer_validate (buffer))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (priv->last_pt != gst_rtp_buffer_get_payload_type (buffer)) {
GstCaps *caps;
priv->last_pt = gst_rtp_buffer_get_payload_type (buffer);
/* reset clock-rate so that we get a new one */
priv->clock_rate = -1;
/* Try to get the clock-rate from the caps first if we can. If there are no
* caps we must fire the signal to get the clock-rate. */
if ((caps = GST_BUFFER_CAPS (buffer))) {
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
}
}
if (priv->clock_rate == -1) {
guint8 pt;
/* no clock rate given on the caps, try to get one with the signal */
pt = gst_rtp_buffer_get_payload_type (buffer);
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
if (priv->clock_rate == -1)
goto not_negotiated;
}
/* take the timestamp of the buffer. This is the time when the packet was
* received and is used to calculate jitter and clock skew. We will adjust
* this timestamp with the smoothed value after processing it in the
* jitterbuffer. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* bring to running time */
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
seqnum = gst_rtp_buffer_get_seq (buffer);
GST_DEBUG_OBJECT (jitterbuffer,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
/* don't accept more data on EOS */
if (priv->eos)
goto have_eos;
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (priv->last_popped_seqnum != -1) {
gint gap;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
if (gap <= 0) {
/* priv->last_popped_seqnum >= seqnum, this packet is too late or the
* sender might have been restarted with different seqnum. */
if (gap < -100) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
} else {
goto too_late;
}
} else {
/* priv->last_popped_seqnum < seqnum, this is a new packet */
if (gap > 3000) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
}
}
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. We can only do this when there actually is a latency. When no
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts) {
GstBuffer *old_buf;
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
seqnum);
old_buf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (old_buf);
}
}
/* now insert the packet into the queue in sorted order. This function returns
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
priv->clock_rate, &tail))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
if (priv->waiting)
JBUF_SIGNAL (priv);
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (priv->clock_id && tail) {
GST_DEBUG_OBJECT (jitterbuffer,
"Unscheduling waiting buffer, new tail buffer");
gst_clock_id_unschedule (priv->clock_id);
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets",
seqnum, rtp_jitter_buffer_num_packets (priv->jbuf));
finished:
JBUF_UNLOCK (priv);
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
not_negotiated:
{
GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
out_flushing:
{
ret = priv->srcresult;
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto finished;
}
have_eos:
{
ret = GST_FLOW_UNEXPECTED;
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
GstFlowReturn
gst_base_video_parse_finish_frame (GstBaseVideoParse * base_video_parse)
{
GstVideoFrame *frame = base_video_parse->current_frame;
GstBuffer *buffer;
GstBaseVideoParseClass *base_video_parse_class;
GstFlowReturn ret;
GST_DEBUG ("finish_frame");
base_video_parse_class = GST_BASE_VIDEO_PARSE_GET_CLASS (base_video_parse);
buffer = gst_adapter_take_buffer (base_video_parse->output_adapter,
gst_adapter_available (base_video_parse->output_adapter));
if (frame->is_sync_point) {
base_video_parse->timestamp_offset = base_video_parse->last_timestamp -
gst_util_uint64_scale (frame->presentation_frame_number,
base_video_parse->state.fps_d * GST_SECOND,
base_video_parse->state.fps_n);
base_video_parse->distance_from_sync = 0;
}
frame->distance_from_sync = base_video_parse->distance_from_sync;
base_video_parse->distance_from_sync++;
frame->presentation_timestamp =
gst_base_video_parse_get_timestamp (base_video_parse,
frame->presentation_frame_number);
frame->presentation_duration =
gst_base_video_parse_get_timestamp (base_video_parse,
frame->presentation_frame_number + 1) - frame->presentation_timestamp;
frame->decode_timestamp =
gst_base_video_parse_get_timestamp (base_video_parse,
frame->decode_frame_number);
GST_BUFFER_TIMESTAMP (buffer) = frame->presentation_timestamp;
GST_BUFFER_DURATION (buffer) = frame->presentation_duration;
if (frame->decode_frame_number < 0) {
GST_BUFFER_OFFSET (buffer) = 0;
} else {
GST_BUFFER_OFFSET (buffer) = frame->decode_timestamp;
}
GST_BUFFER_OFFSET_END (buffer) = GST_CLOCK_TIME_NONE;
GST_DEBUG ("pts %" GST_TIME_FORMAT,
GST_TIME_ARGS (frame->presentation_timestamp));
GST_DEBUG ("dts %" GST_TIME_FORMAT, GST_TIME_ARGS (frame->decode_timestamp));
GST_DEBUG ("dist %d", frame->distance_from_sync);
if (frame->is_sync_point) {
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
} else {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
}
frame->src_buffer = buffer;
ret = base_video_parse_class->shape_output (base_video_parse, frame);
gst_base_video_parse_free_frame (base_video_parse->current_frame);
/* create new frame */
base_video_parse->current_frame =
gst_base_video_parse_new_frame (base_video_parse);
return ret;
}
static GstMessage *
update_rms_from_buffer (GstVideoFrameAudioLevel * self, GstBuffer * inbuf)
{
GstMapInfo map;
guint8 *in_data;
gsize in_size;
gdouble CS;
guint i;
guint num_frames, frames;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
gint channels, rate, bps;
GValue v = G_VALUE_INIT;
GValue va = G_VALUE_INIT;
GValueArray *a;
GstStructure *s;
GstMessage *msg;
GstClockTime duration, running_time;
channels = GST_AUDIO_INFO_CHANNELS (&self->ainfo);
bps = GST_AUDIO_INFO_BPS (&self->ainfo);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
gst_buffer_map (inbuf, &map, GST_MAP_READ);
in_data = map.data;
in_size = map.size;
num_int_samples = in_size / bps;
GST_LOG_OBJECT (self, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)));
g_return_val_if_fail (num_int_samples % channels == 0, NULL);
num_frames = num_int_samples / channels;
frames = num_frames;
duration = GST_FRAMES_TO_CLOCK_TIME (frames, rate);
if (num_frames > 0) {
for (i = 0; i < channels; ++i) {
self->process (in_data + (bps * i), num_int_samples, channels, &CS);
GST_LOG_OBJECT (self,
"[%d]: cumulative squares %lf, over %d samples/%d channels",
i, CS, num_int_samples, channels);
self->CS[i] += CS;
}
in_data += num_frames * bps;
self->total_frames += num_frames;
}
running_time =
self->first_time + gst_util_uint64_scale (self->total_frames, GST_SECOND,
rate);
a = g_value_array_new (channels);
s = gst_structure_new ("videoframe-audiolevel", "running-time", G_TYPE_UINT64,
running_time, "duration", G_TYPE_UINT64, duration, NULL);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_init (&va, G_TYPE_VALUE_ARRAY);
for (i = 0; i < channels; i++) {
gdouble rms;
if (frames == 0 || self->CS[i] == 0) {
rms = 0; /* empty buffer */
} else {
rms = sqrt (self->CS[i] / frames);
}
self->CS[i] = 0.0;
g_value_set_double (&v, rms);
g_value_array_append (a, &v);
}
g_value_take_boxed (&va, a);
gst_structure_take_value (s, "rms", &va);
msg = gst_message_new_element (GST_OBJECT (self), s);
gst_buffer_unmap (inbuf, &map);
return msg;
}
static GstFlowReturn mfw_gst_vpuenc_chain(GstPad * pad, GstBuffer * buffer)
{
GstVPU_Enc *vpu_enc = NULL;
GstFlowReturn retval = GST_FLOW_OK;
GstCaps *src_caps;
GstBuffer *outbuffer;
gint i = 0;
int ret;
struct pollfd pollfd;
unsigned long type = V4L2_BUF_TYPE_VIDEO_OUTPUT;
GST_DEBUG(__func__);
vpu_enc = MFW_GST_VPU_ENC(GST_PAD_PARENT(pad));
if (vpu_enc->init == FALSE) {
retval = mfw_gst_vpuenc_init_encoder(pad, vpu_enc->memory);
if (retval != GST_FLOW_OK)
return retval;
printf("VPU ENC initialised\n");
}
i = 0;
if (vpu_enc->memory == V4L2_MEMORY_USERPTR) {
for (i = 0; i < NUM_BUFFERS; i++) {
if (vpu_enc->buf_v4l2[i].m.userptr == (long int)GST_BUFFER_DATA (buffer))
break;
}
if (i == NUM_BUFFERS) {
for (i = 0; i < NUM_BUFFERS; i++) {
if (!vpu_enc->buf_v4l2[i].m.userptr)
break;
}
}
i = i % NUM_BUFFERS;
}
if (i == NUM_BUFFERS) {
printf("NO BUFFER AVAILABLE\n");
return GST_FLOW_ERROR;
}
if (!buffer)
return GST_FLOW_OK;
if (vpu_enc->memory == V4L2_MEMORY_MMAP) {
/* copy the input Frame into the allocated buffer */
memcpy(vpu_enc->buf_data[i], GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer));
gst_buffer_unref(buffer);
} else {
vpu_enc->buf_v4l2[i].m.userptr = (long int)GST_BUFFER_DATA (buffer);
vpu_enc->buf_v4l2[i].length = GST_BUFFER_SIZE (buffer);
}
pollfd.fd = vpu_enc->vpu_fd;
pollfd.events = POLLIN | POLLOUT;
ret = ioctl(vpu_enc->vpu_fd, VIDIOC_QBUF, &vpu_enc->buf_v4l2[i]);
if (ret) {
if (vpu_enc->memory == V4L2_MEMORY_USERPTR) {
/* fallback to mmap */
vpu_enc->init = FALSE;
vpu_enc->memory = V4L2_MEMORY_MMAP;
GST_WARNING("mfw_gst_vpuenc_chain: fallback to mmap");
return mfw_gst_vpuenc_chain(pad, buffer);
}
GST_ERROR("VIDIOC_QBUF failed: %s\n", strerror(errno));
return GST_FLOW_ERROR;
}
if (!vpu_enc->once) {
retval = ioctl(vpu_enc->vpu_fd, VIDIOC_STREAMON, &type);
if (retval) {
printf("streamon failed with %d", retval);
return GST_FLOW_ERROR;
}
vpu_enc->once = 1;
}
ret = ioctl(vpu_enc->vpu_fd, VIDIOC_DQBUF, &vpu_enc->buf_v4l2[0]);
if (ret) {
GST_ERROR("VIDIOC_DQBUF failed: %s\n", strerror(errno));
return GST_FLOW_ERROR;
}
if (vpu_enc->memory == V4L2_MEMORY_USERPTR) {
gst_buffer_unref(buffer);
}
src_caps = GST_PAD_CAPS(vpu_enc->srcpad);
retval = gst_pad_alloc_buffer_and_set_caps(vpu_enc->srcpad,
0, 1024 * 1024, src_caps, &outbuffer);
if (retval != GST_FLOW_OK) {
GST_ERROR("Allocating buffer failed with %d", ret);
return retval;
}
ret = read(vpu_enc->vpu_fd, GST_BUFFER_DATA(outbuffer), 1024 * 1024);
if (ret < 0) {
printf("read failed: %s\n", strerror(errno));
return GST_FLOW_ERROR;
}
GST_BUFFER_SIZE(outbuffer) = ret;
GST_BUFFER_TIMESTAMP(outbuffer) = gst_util_uint64_scale(vpu_enc->encoded_frames,
1 * GST_SECOND,
vpu_enc->framerate);
vpu_enc->encoded_frames++;
GST_DEBUG_OBJECT(vpu_enc, "frame encoded : %lld ts = %" GST_TIME_FORMAT,
vpu_enc->encoded_frames,
GST_TIME_ARGS(GST_BUFFER_TIMESTAMP(outbuffer)));
retval = gst_pad_push(vpu_enc->srcpad, outbuffer);
if (retval != GST_FLOW_OK) {
GST_ERROR("Pushing Output onto the source pad failed with %d \n",
retval);
}
return retval;
}
/* Get a DV frame, chop it up in pieces, and push the pieces to the RTP layer.
*/
static GstFlowReturn
gst_rtp_dv_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPDVPay *rtpdvpay;
guint max_payload_size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
gint hdrlen;
gsize size;
GstMapInfo map;
guint8 *data;
guint8 *dest;
guint filled;
GstRTPBuffer rtp = { NULL, };
rtpdvpay = GST_RTP_DV_PAY (basepayload);
hdrlen = gst_rtp_buffer_calc_header_len (0);
/* DV frames are made up from a bunch of DIF blocks. DIF blocks are 80 bytes
* each, and we should put an integral number of them in each RTP packet.
* Therefore, we round the available room down to the nearest multiple of 80.
*
* The available room is just the packet MTU, minus the RTP header length. */
max_payload_size = ((GST_RTP_BASE_PAYLOAD_MTU (rtpdvpay) - hdrlen) / 80) * 80;
/* The length of the buffer to transmit. */
if (!gst_buffer_map (buffer, &map, GST_MAP_READ)) {
GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
(NULL), ("Failed to map buffer"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (rtpdvpay,
"DV RTP payloader got buffer of %" G_GSIZE_FORMAT
" bytes, splitting in %u byte " "payload fragments, at time %"
GST_TIME_FORMAT, size, max_payload_size,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
if (!rtpdvpay->negotiated) {
gst_dv_pay_negotiate (rtpdvpay, data, size);
/* if we have not yet scanned the stream for its type, do so now */
rtpdvpay->negotiated = TRUE;
}
outbuf = NULL;
dest = NULL;
filled = 0;
/* while we have a complete DIF chunks left */
while (size >= 80) {
/* Allocate a new buffer, set the timestamp */
if (outbuf == NULL) {
outbuf = gst_rtp_buffer_new_allocate (max_payload_size, 0, 0);
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buffer);
if (!gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp)) {
gst_buffer_unref (outbuf);
GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
(NULL), ("Failed to map RTP buffer"));
ret = GST_FLOW_ERROR;
goto beach;
}
dest = gst_rtp_buffer_get_payload (&rtp);
filled = 0;
}
/* inspect the DIF chunk, if we don't need to include it, skip to the next one. */
if (include_dif (rtpdvpay, data)) {
/* copy data in packet */
memcpy (dest, data, 80);
dest += 80;
filled += 80;
}
/* go to next dif chunk */
size -= 80;
data += 80;
/* push out the buffer if the next one would exceed the max packet size or
* when we are pushing the last packet */
if (filled + 80 > max_payload_size || size < 80) {
if (size < 160) {
guint hlen;
/* set marker */
gst_rtp_buffer_set_marker (&rtp, TRUE);
/* shrink buffer to last packet */
hlen = gst_rtp_buffer_get_header_len (&rtp);
gst_rtp_buffer_set_packet_len (&rtp, hlen + filled);
}
/* Push out the created piece, and check for errors. */
gst_rtp_buffer_unmap (&rtp);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
outbuf = NULL;
}
}
beach:
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return ret;
}
static GstFlowReturn
gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
baseaudiopayload, GstBuffer * buffer)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstClockTime timestamp;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
payload_len = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) {
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
}
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
GstBufferListIterator *it;
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, buffer);
gst_buffer_list_iterator_free (it);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_basertppayload_push_list (basepayload, list);
} else {
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, GST_BUFFER_DATA (buffer), payload_len);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}