static GstFlowReturn
gst_ogg_avi_parse_push_packet (GstOggAviParse * ogg, ogg_packet * packet)
{
GstBuffer *buffer;
GstFlowReturn result;
/* allocate space for header and body */
buffer = gst_buffer_new_and_alloc (packet->bytes);
memcpy (GST_BUFFER_DATA (buffer), packet->packet, packet->bytes);
GST_LOG_OBJECT (ogg, "created buffer %p from page", buffer);
GST_BUFFER_OFFSET_END (buffer) = packet->granulepos;
if (ogg->discont) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
ogg->discont = FALSE;
}
result = gst_pad_push (ogg->srcpad, buffer);
return result;
}
static void
gst_identity_update_last_message_for_buffer (GstIdentity * identity,
const gchar * action, GstBuffer * buf)
{
gchar ts_str[64], dur_str[64];
GST_OBJECT_LOCK (identity);
g_free (identity->last_message);
identity->last_message = g_strdup_printf ("%s ******* (%s:%s)i "
"(%u bytes, timestamp: %s, duration: %s, offset: %" G_GINT64_FORMAT ", "
"offset_end: % " G_GINT64_FORMAT ", flags: %d) %p", action,
GST_DEBUG_PAD_NAME (GST_BASE_TRANSFORM_CAST (identity)->sinkpad),
GST_BUFFER_SIZE (buf),
print_pretty_time (ts_str, sizeof (ts_str), GST_BUFFER_TIMESTAMP (buf)),
print_pretty_time (dur_str, sizeof (dur_str), 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);
}
static void
gst_multi_file_sink_post_message (GstMultiFileSink * multifilesink,
GstBuffer * buffer, const char *filename)
{
if (multifilesink->post_messages) {
GstClockTime duration, timestamp;
GstClockTime running_time, stream_time;
guint64 offset, offset_end;
GstStructure *s;
GstSegment *segment;
GstFormat format;
segment = &GST_BASE_SINK (multifilesink)->segment;
format = segment->format;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
offset = GST_BUFFER_OFFSET (buffer);
offset_end = GST_BUFFER_OFFSET_END (buffer);
running_time = gst_segment_to_running_time (segment, format, timestamp);
stream_time = gst_segment_to_stream_time (segment, format, timestamp);
s = gst_structure_new ("GstMultiFileSink",
"filename", G_TYPE_STRING, filename,
"index", G_TYPE_INT, multifilesink->index,
"timestamp", G_TYPE_UINT64, timestamp,
"stream-time", G_TYPE_UINT64, stream_time,
"running-time", G_TYPE_UINT64, running_time,
"duration", G_TYPE_UINT64, duration,
"offset", G_TYPE_UINT64, offset,
"offset-end", G_TYPE_UINT64, offset_end, NULL);
gst_element_post_message (GST_ELEMENT_CAST (multifilesink),
gst_message_new_element (GST_OBJECT_CAST (multifilesink), s));
}
}
static GstFlowReturn
gst_kate_parse_push_buffer (GstKateParse * parse, GstBuffer * buf,
gint64 granulepos)
{
GST_LOG_OBJECT (parse, "granulepos %16" G_GINT64_MODIFIER "x", granulepos);
if (granulepos < 0) {
/* packets coming not from Ogg won't have a granpos in the offset end,
so we have to synthesize one here - only problem is we don't know
the backlink - pretend there's none for now */
GST_INFO_OBJECT (parse, "No granulepos on buffer, synthesizing one");
granulepos =
kate_duration_granule (&parse->ki,
GST_BUFFER_TIMESTAMP (buf) /
(double) GST_SECOND) << kate_granule_shift (&parse->ki);
}
GST_BUFFER_OFFSET (buf) =
kate_granule_time (&parse->ki, granulepos) * GST_SECOND;
GST_BUFFER_OFFSET_END (buf) = granulepos;
GST_BUFFER_TIMESTAMP (buf) = GST_BUFFER_OFFSET (buf);
gst_buffer_set_caps (buf, GST_PAD_CAPS (parse->srcpad));
return gst_pad_push (parse->srcpad, buf);
}
static GstFlowReturn
send_buffers_before(AudioTrim *filter, gint64 before)
{
GstFlowReturn ret = GST_FLOW_OK;
GList *b = filter->buffers;
while(b) {
GstBuffer *buf = b->data;
b->data = NULL;
if ((gint64)GST_BUFFER_OFFSET(buf) >= before) {
gst_buffer_unref(buf);
} else {
if ((gint64)GST_BUFFER_OFFSET_END(buf) > before) {
GstBuffer *head = buffer_head(filter, buf, before);
gst_buffer_unref(buf);
buf = head;
}
ret = gst_pad_push(filter->srcpad, buf);
if (ret != GST_FLOW_OK) break;
}
b = g_list_next(b);
}
release_buffers(filter);
return ret;
}
static GstFlowReturn
send_buffers_after(AudioTrim *filter, gint64 after)
{
GstFlowReturn ret = GST_FLOW_OK;
GList *b = filter->buffers;
while(b) {
GstBuffer *buf = b->data;
b->data = NULL;
if ((gint64)GST_BUFFER_OFFSET_END(buf) <= after) {
gst_buffer_unref(buf);
} else {
if ((gint64)GST_BUFFER_OFFSET(buf) < after) {
GstBuffer *tail = buffer_tail(filter, buf, after);
gst_buffer_unref(buf);
buf = tail;
}
ret = gst_pad_push(filter->srcpad, buf);
if (ret != GST_FLOW_OK) break;
}
b = g_list_next(b);
}
release_buffers(filter);
return ret;
}
void ofxGstRTPServer::newOscMsg(ofxOscMessage & msg, GstClockTime timestamp){
if(!appSrcOsc) return;
GstClockTime now = timestamp;
if(!oscAutoTimestamp){
if(now==GST_CLOCK_TIME_NONE){
now = getTimeStamp();
}
if(firstOscFrame){
prevTimestampOsc = now;
firstOscFrame = false;
return;
}
}
PooledOscPacket * pooledOscPkg = oscPacketPool.newBuffer();
appendMessage(msg,pooledOscPkg->packet);
GstBuffer * buffer = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY,(void*)pooledOscPkg->compressedData(),pooledOscPkg->compressedSize(),0,pooledOscPkg->compressedSize(),pooledOscPkg,(GDestroyNotify)&ofxOscPacketPool::relaseBuffer);
if(!oscAutoTimestamp){
GST_BUFFER_OFFSET(buffer) = numFrameOsc++;
GST_BUFFER_OFFSET_END(buffer) = numFrameOsc;
GST_BUFFER_DTS (buffer) = now;
GST_BUFFER_PTS (buffer) = now;
GST_BUFFER_DURATION(buffer) = now-prevTimestampOsc;
prevTimestampOsc = now;
}
GstFlowReturn flow_return = gst_app_src_push_buffer((GstAppSrc*)appSrcOsc, buffer);
if (flow_return != GST_FLOW_OK) {
ofLogError() << "error pushing osc buffer: flow_return was " << flow_return;
}
}
void
gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER_RATE (self);
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
gint outsize, outsamples;
GstMapInfo map;
guint8 *in, *out;
if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = self->nsamples_in - (self->nsamples_out - self->latency);
if (outsamples <= 0) {
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
return;
}
outsize = outsamples * channels * bps;
if (!self->fft || self->low_latency) {
gint64 diffsize, diffsamples;
/* Process the difference between latency and residue length samples
* to start at the actual data instead of starting at the zeros before
* when we only got one buffer smaller than latency */
diffsamples =
((gint64) self->latency) - ((gint64) self->buffer_fill) / channels;
if (diffsamples > 0) {
diffsize = diffsamples * channels * bps;
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->nsamples_out += self->process (self, in, out, diffsamples);
g_free (in);
g_free (out);
}
outbuf = gst_buffer_new_and_alloc (outsize);
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
self->nsamples_out += self->process (self, in, map.data, outsamples);
gst_buffer_unmap (outbuf, &map);
g_free (in);
} else {
guint gensamples = 0;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &map, GST_MAP_READWRITE);
while (gensamples < outsamples) {
guint step_insamples = self->block_length - self->buffer_fill;
guint8 *zeroes = g_new0 (guint8, step_insamples * channels * bps);
guint8 *out = g_new (guint8, self->block_length * channels * bps);
guint step_gensamples;
step_gensamples = self->process (self, zeroes, out, step_insamples);
g_free (zeroes);
memcpy (map.data + gensamples * bps, out, MIN (step_gensamples,
outsamples - gensamples) * bps);
gensamples += MIN (step_gensamples, outsamples - gensamples);
g_free (out);
}
self->nsamples_out += gensamples;
gst_buffer_unmap (outbuf, &map);
}
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->start_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_TIMESTAMP (outbuf) +=
gst_util_uint64_scale_int (self->nsamples_out - outsamples -
self->latency, GST_SECOND, rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) =
self->start_off + self->nsamples_out - outsamples - self->latency;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + outsamples;
}
GST_DEBUG_OBJECT (self,
"Pushing residue buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
gst_buffer_get_size (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), outsamples);
res = gst_pad_push (GST_BASE_TRANSFORM_CAST (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
self->buffer_fill = 0;
}
static void
do_perfect_stream_test (guint rate, guint width, gdouble drop_probability,
gdouble inject_probability)
{
GstElement *pipe, *src, *conv, *filter, *injector, *audiorate, *sink;
GstMessage *msg;
GstCaps *caps;
GstPad *srcpad;
GList *l, *bufs = NULL;
GstClockTime next_time = GST_CLOCK_TIME_NONE;
guint64 next_offset = GST_BUFFER_OFFSET_NONE;
caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT,
rate, "width", G_TYPE_INT, width, NULL);
GST_INFO ("-------- drop=%.0f%% caps = %" GST_PTR_FORMAT " ---------- ",
drop_probability * 100.0, caps);
g_assert (drop_probability >= 0.0 && drop_probability <= 1.0);
g_assert (inject_probability >= 0.0 && inject_probability <= 1.0);
g_assert (width > 0 && (width % 8) == 0);
pipe = gst_pipeline_new ("pipeline");
fail_unless (pipe != NULL);
src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
fail_unless (src != NULL);
g_object_set (src, "num-buffers", 100, NULL);
conv = gst_element_factory_make ("audioconvert", "audioconvert");
fail_unless (conv != NULL);
filter = gst_element_factory_make ("capsfilter", "capsfilter");
fail_unless (filter != NULL);
g_object_set (filter, "caps", caps, NULL);
injector_inject_probability = inject_probability;
injector = GST_ELEMENT (g_object_new (test_injector_get_type (), NULL));
srcpad = gst_element_get_pad (injector, "src");
fail_unless (srcpad != NULL);
gst_pad_add_buffer_probe (srcpad, G_CALLBACK (probe_cb), &drop_probability);
gst_object_unref (srcpad);
audiorate = gst_element_factory_make ("audiorate", "audiorate");
fail_unless (audiorate != NULL);
sink = gst_element_factory_make ("fakesink", "fakesink");
fail_unless (sink != NULL);
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &bufs);
gst_bin_add_many (GST_BIN (pipe), src, conv, filter, injector, audiorate,
sink, NULL);
gst_element_link_many (src, conv, filter, injector, audiorate, sink, NULL);
fail_unless_equals_int (gst_element_set_state (pipe, GST_STATE_PLAYING),
GST_STATE_CHANGE_ASYNC);
fail_unless_equals_int (gst_element_get_state (pipe, NULL, NULL, -1),
GST_STATE_CHANGE_SUCCESS);
msg = gst_bus_poll (GST_ELEMENT_BUS (pipe),
GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
fail_unless_equals_string (GST_MESSAGE_TYPE_NAME (msg), "eos");
for (l = bufs; l != NULL; l = l->next) {
GstBuffer *buf = GST_BUFFER (l->data);
guint num_samples;
fail_unless (GST_BUFFER_TIMESTAMP_IS_VALID (buf));
fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_END_IS_VALID (buf));
GST_LOG ("buffer: ts=%" GST_TIME_FORMAT ", end_ts=%" GST_TIME_FORMAT
" off=%" G_GINT64_FORMAT ", end_off=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
if (GST_CLOCK_TIME_IS_VALID (next_time)) {
fail_unless_equals_uint64 (next_time, GST_BUFFER_TIMESTAMP (buf));
}
if (next_offset != GST_BUFFER_OFFSET_NONE) {
fail_unless_equals_uint64 (next_offset, GST_BUFFER_OFFSET (buf));
}
/* check buffer size for sanity */
fail_unless_equals_int (GST_BUFFER_SIZE (buf) % (width / 8), 0);
/* check there is actually as much data as there should be */
num_samples = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
fail_unless_equals_int (GST_BUFFER_SIZE (buf), num_samples * (width / 8));
next_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
next_offset = GST_BUFFER_OFFSET_END (buf);
}
gst_message_unref (msg);
gst_element_set_state (pipe, GST_STATE_NULL);
gst_object_unref (pipe);
g_list_foreach (bufs, (GFunc) gst_mini_object_unref, NULL);
g_list_free (bufs);
gst_caps_unref (caps);
}
static GstFlowReturn
gst_aiff_parse_stream_data (GstAiffParse * aiff)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
G_GINT64_FORMAT, aiff->offset, aiff->end_offset, aiff->dataleft);
/* Get the next n bytes and output them */
if (aiff->dataleft == 0 || aiff->dataleft < aiff->bytes_per_sample)
goto found_eos;
/* scale the amount of data by the segment rate so we get equal
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (aiff->dataleft),
MAX_BUFFER_SIZE * aiff->segment.abs_rate);
if (desired >= aiff->bytes_per_sample && aiff->bytes_per_sample > 0)
desired -= (desired % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "Fetching %" G_GINT64_FORMAT " bytes of data "
"from the sinkpad", desired);
if (aiff->streaming) {
guint avail = gst_adapter_available (aiff->adapter);
if (avail < desired) {
GST_LOG_OBJECT (aiff, "Got only %d bytes of data from the sinkpad",
avail);
return GST_FLOW_OK;
}
buf = gst_adapter_take_buffer (aiff->adapter, desired);
} else {
if ((res = gst_pad_pull_range (aiff->sinkpad, aiff->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
}
/* If we have a pending close/start segment, send it now. */
if (G_UNLIKELY (aiff->close_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->close_segment);
aiff->close_segment = NULL;
}
if (G_UNLIKELY (aiff->start_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->start_segment);
aiff->start_segment = NULL;
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions in bytes */
pos = aiff->offset - aiff->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
GST_BUFFER_OFFSET (buf) = pos / aiff->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / aiff->bytes_per_sample;
if (aiff->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) aiff->bps);
next_timestamp =
uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) aiff->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_TIME, next_timestamp);
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
* other positions and durations have unknown timestamp. */
if (pos == 0)
timestamp = 0;
else
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_BYTES, nextpos);
}
if (aiff->discont) {
GST_DEBUG_OBJECT (aiff, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
aiff->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
gst_buffer_set_caps (buf, aiff->caps);
GST_LOG_OBJECT (aiff,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (aiff->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < aiff->dataleft) {
aiff->offset += obtained;
aiff->dataleft -= obtained;
} else {
aiff->offset += aiff->dataleft;
aiff->dataleft = 0;
}
/* Iterate until need more data, so adapter size won't grow */
if (aiff->streaming) {
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, aiff->offset,
aiff->end_offset);
goto iterate_adapter;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (aiff, "found EOS");
return GST_FLOW_UNEXPECTED;
}
pull_error:
{
/* check if we got EOS */
if (res == GST_FLOW_UNEXPECTED)
goto found_eos;
GST_WARNING_OBJECT (aiff,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, aiff->dataleft);
return res;
}
push_error:
{
GST_INFO_OBJECT (aiff,
"Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
GST_DEBUG_PAD_NAME (aiff->srcpad), gst_flow_get_name (res),
gst_pad_is_linked (aiff->srcpad));
return res;
}
}
static GstFlowReturn
gst_split_file_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
GstBuffer ** buffer)
{
GstSplitFileSrc *src = GST_SPLIT_FILE_SRC (basesrc);
GstFilePart cur_part;
GInputStream *stream;
GCancellable *cancel;
GSeekable *seekable;
GstBuffer *buf;
GError *err = NULL;
guint64 read_offset;
guint8 *data;
guint to_read;
cur_part = src->parts[src->cur_part];
if (offset < cur_part.start || offset > cur_part.stop) {
if (!gst_split_file_src_find_part_for_offset (src, offset, &src->cur_part))
return GST_FLOW_UNEXPECTED;
cur_part = src->parts[src->cur_part];
}
GST_LOG_OBJECT (src, "current part: %u (%" G_GUINT64_FORMAT " - "
"%" G_GUINT64_FORMAT ", %s)", src->cur_part, cur_part.start,
cur_part.stop, cur_part.path);
buf = gst_buffer_new_and_alloc (size);
GST_BUFFER_OFFSET (buf) = offset;
data = GST_BUFFER_DATA (buf);
cancel = src->cancellable;
while (size > 0) {
guint64 bytes_to_end_of_part;
gsize read = 0;
/* we want the offset into the file part */
read_offset = offset - cur_part.start;
GST_LOG ("Reading part %03u from offset %" G_GUINT64_FORMAT " (%s)",
src->cur_part, read_offset, cur_part.path);
/* FIXME: only seek when needed (hopefully gio is smart) */
seekable = G_SEEKABLE (cur_part.stream);
if (!g_seekable_seek (seekable, read_offset, G_SEEK_SET, cancel, &err))
goto seek_failed;
GST_LOG_OBJECT (src, "now: %" G_GUINT64_FORMAT, g_seekable_tell (seekable));
bytes_to_end_of_part = (cur_part.stop - cur_part.start) + 1 - read_offset;
to_read = MIN (size, bytes_to_end_of_part);
GST_LOG_OBJECT (src, "reading %u bytes from part %u (bytes to end of "
"part: %u)", to_read, src->cur_part, (guint) bytes_to_end_of_part);
stream = G_INPUT_STREAM (cur_part.stream);
/* NB: we won't try to read beyond EOF */
if (!g_input_stream_read_all (stream, data, to_read, &read, cancel, &err))
goto read_failed;
GST_LOG_OBJECT (src, "read %u bytes", (guint) read);
data += read;
size -= read;
offset += read;
/* are we done? */
if (size == 0)
break;
GST_LOG_OBJECT (src, "%u bytes left to read for this chunk", size);
/* corner case, this should never really happen (assuming basesrc clips
* requests beyond the file size) */
if (read < to_read) {
if (src->cur_part == src->num_parts - 1) {
/* last file part, stop reading and truncate buffer */
GST_BUFFER_SIZE (buf) = offset - GST_BUFFER_OFFSET (buf);
break;
} else {
goto file_part_changed;
}
}
++src->cur_part;
cur_part = src->parts[src->cur_part];
}
GST_BUFFER_OFFSET_END (buf) = offset;
*buffer = buf;
GST_LOG_OBJECT (src, "read %u bytes into buf %p", GST_BUFFER_SIZE (buf), buf);
return GST_FLOW_OK;
/* ERRORS */
seek_failed:
{
if (err->code == G_IO_ERROR_CANCELLED)
goto cancelled;
GST_ELEMENT_ERROR (src, RESOURCE, SEEK, (NULL),
("Seek to %" G_GUINT64_FORMAT " in %s failed", read_offset,
cur_part.path));
g_error_free (err);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
read_failed:
{
if (err->code == G_IO_ERROR_CANCELLED)
goto cancelled;
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("%s", err->message),
("Read from %" G_GUINT64_FORMAT " in %s failed", read_offset,
cur_part.path));
g_error_free (err);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
file_part_changed:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ,
("Read error while reading file part %s", cur_part.path),
("Short read in file part, file may have been modified since start"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
cancelled:
{
GST_DEBUG_OBJECT (src, "I/O operation cancelled from another thread");
g_error_free (err);
gst_buffer_unref (buf);
return GST_FLOW_WRONG_STATE;
}
}
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);
gsize size;
size = gst_buffer_get_size (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)
goto error_after;
}
if (identity->drop_probability > 0.0) {
if ((gfloat) (1.0 * rand () / (RAND_MAX)) < identity->drop_probability)
goto dropped;
}
if (identity->dump) {
GstMapInfo info;
gst_buffer_map (buf, &info, GST_MAP_READ);
gst_util_dump_mem (info.data, info.size);
gst_buffer_unmap (buf, &info);
}
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "chain", buf, size);
}
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) = size * 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 */
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_EOS;
}
GST_OBJECT_UNLOCK (identity);
}
identity->offset += size;
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;
/* ERRORS */
error_after:
{
GST_ELEMENT_ERROR (identity, CORE, FAILED,
(_("Failed after iterations as requested.")), (NULL));
return GST_FLOW_ERROR;
}
dropped:
{
if (!identity->silent) {
gst_identity_update_last_message_for_buffer (identity, "dropping", buf,
size);
}
/* return DROPPED to basetransform. */
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
static gboolean
gst_ks_video_src_timestamp_buffer (GstKsVideoSrc * self, GstBuffer * buf,
GstClockTime presentation_time)
{
GstKsVideoSrcPrivate *priv = GST_KS_VIDEO_SRC_GET_PRIVATE (self);
GstClockTime duration;
GstClock *clock;
GstClockTime timestamp;
duration = gst_ks_video_device_get_duration (priv->device);
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL) {
gst_object_ref (clock);
timestamp = GST_ELEMENT (self)->base_time;
if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
if (presentation_time > GST_ELEMENT (self)->base_time)
presentation_time -= GST_ELEMENT (self)->base_time;
else
presentation_time = 0;
}
} else {
timestamp = GST_CLOCK_TIME_NONE;
}
GST_OBJECT_UNLOCK (self);
if (clock != NULL) {
/* The time according to the current clock */
timestamp = gst_clock_get_time (clock) - timestamp;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
if (GST_CLOCK_TIME_IS_VALID (presentation_time)) {
/*
* We don't use this for anything yet, need to ponder how to deal
* with pins that use an internal clock and timestamp from 0.
*/
GstClockTimeDiff diff = GST_CLOCK_DIFF (presentation_time, timestamp);
GST_DEBUG_OBJECT (self, "diff between gst and driver timestamp: %"
G_GINT64_FORMAT, diff);
}
gst_object_unref (clock);
clock = NULL;
/* Unless it's the first frame, align the current timestamp on a multiple
* of duration since the previous */
if (GST_CLOCK_TIME_IS_VALID (priv->prev_ts)) {
GstClockTime delta;
guint delta_remainder, delta_offset;
/* REVISIT: I've seen this happen with the GstSystemClock on Windows,
* scary... */
if (timestamp < priv->prev_ts) {
GST_INFO_OBJECT (self, "clock is ticking backwards");
return FALSE;
}
/* Round to a duration boundary */
delta = timestamp - priv->prev_ts;
delta_remainder = delta % duration;
if (delta_remainder < duration / 3)
timestamp -= delta_remainder;
else
timestamp += duration - delta_remainder;
/* How many frames are we off then? */
delta = timestamp - priv->prev_ts;
delta_offset = delta / duration;
if (delta_offset == 1) /* perfect */
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
else if (delta_offset > 1) {
guint lost = delta_offset - 1;
GST_INFO_OBJECT (self, "lost %d frame%s, setting discont flag",
lost, (lost > 1) ? "s" : "");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
} else if (delta_offset == 0) { /* overproduction, skip this frame */
GST_INFO_OBJECT (self, "skipping frame");
return FALSE;
}
priv->offset += delta_offset;
}
priv->prev_ts = timestamp;
}
GST_BUFFER_OFFSET (buf) = priv->offset;
GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET (buf) + 1;
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
return TRUE;
}
GstFlowReturn
gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
GstVideoFrame * frame)
{
GstBaseVideoDecoderClass *base_video_decoder_class;
GstBuffer *src_buffer;
GST_DEBUG ("finish frame");
base_video_decoder_class =
GST_BASE_VIDEO_DECODER_GET_CLASS (base_video_decoder);
GST_DEBUG ("finish frame sync=%d pts=%" GST_TIME_FORMAT,
GST_VIDEO_FRAME_FLAG_IS_SET (frame, GST_VIDEO_FRAME_FLAG_SYNC_POINT),
GST_TIME_ARGS (frame->presentation_timestamp));
if (GST_CLOCK_TIME_IS_VALID (frame->presentation_timestamp)) {
if (frame->presentation_timestamp != base_video_decoder->timestamp_offset) {
GST_DEBUG ("sync timestamp %" GST_TIME_FORMAT " diff %" GST_TIME_FORMAT,
GST_TIME_ARGS (frame->presentation_timestamp),
GST_TIME_ARGS (frame->presentation_timestamp -
base_video_decoder->segment.start));
base_video_decoder->timestamp_offset = frame->presentation_timestamp;
base_video_decoder->field_index = 0;
} else {
/* This case is for one initial timestamp and no others, e.g.,
* filesrc ! decoder ! xvimagesink */
GST_WARNING ("sync timestamp didn't change, ignoring");
frame->presentation_timestamp = GST_CLOCK_TIME_NONE;
}
} else {
if (GST_VIDEO_FRAME_FLAG_IS_SET (frame, GST_VIDEO_FRAME_FLAG_SYNC_POINT)) {
GST_WARNING ("sync point doesn't have timestamp");
if (!GST_CLOCK_TIME_IS_VALID (base_video_decoder->timestamp_offset)) {
GST_WARNING
("No base timestamp. Assuming frames start at segment start");
base_video_decoder->timestamp_offset =
base_video_decoder->segment.start;
base_video_decoder->field_index = 0;
}
}
}
frame->field_index = base_video_decoder->field_index;
base_video_decoder->field_index += frame->n_fields;
if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
frame->presentation_timestamp =
gst_base_video_decoder_get_field_timestamp (base_video_decoder,
frame->field_index);
frame->presentation_duration = GST_CLOCK_TIME_NONE;
frame->decode_timestamp =
gst_base_video_decoder_get_timestamp (base_video_decoder,
frame->decode_frame_number);
}
if (frame->presentation_duration == GST_CLOCK_TIME_NONE) {
frame->presentation_duration =
gst_base_video_decoder_get_field_duration (base_video_decoder,
frame->n_fields);
}
if (GST_CLOCK_TIME_IS_VALID (base_video_decoder->last_timestamp)) {
if (frame->presentation_timestamp < base_video_decoder->last_timestamp) {
GST_WARNING ("decreasing timestamp (%" GST_TIME_FORMAT " < %"
GST_TIME_FORMAT ")", GST_TIME_ARGS (frame->presentation_timestamp),
GST_TIME_ARGS (base_video_decoder->last_timestamp));
}
}
base_video_decoder->last_timestamp = frame->presentation_timestamp;
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) = frame->presentation_timestamp;
GST_BUFFER_DURATION (src_buffer) = frame->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 (frame->presentation_timestamp));
gst_base_video_decoder_set_src_caps (base_video_decoder);
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);
}
EXPORT_C
#endif
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
gint frame_size)
{
GstBuffer *ret;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
guint8 *data;
guint size;
gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
TRUE;
g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
segment->format == GST_FORMAT_DEFAULT, buffer);
g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
/* No timestamp - assume the buffer is completely in the segment */
return buffer;
/* Get copies of the buffer metadata to change later.
* Calculate the missing values for the calculations,
* they won't be changed later though. */
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
offset = GST_BUFFER_OFFSET (buffer);
} else {
change_offset = FALSE;
offset = 0;
}
if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
offset_end = offset + size / frame_size;
}
if (segment->format == GST_FORMAT_TIME) {
/* Handle clipping for GST_FORMAT_TIME */
gint64 start, stop, cstart, cstop, diff;
start = timestamp;
stop = timestamp + duration;
if (gst_segment_clip (segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
timestamp = cstart;
if (change_duration)
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
data += diff * frame_size;
size -= diff * frame_size;
}
diff = stop - cstop;
if (diff > 0) {
/* duration is always valid if stop is valid */
duration -= diff;
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
size -= diff * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
} else {
/* Handle clipping for GST_FORMAT_DEFAULT */
gint64 start, stop, cstart, cstop, diff;
g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);
start = offset;
stop = offset_end;
if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
offset = cstart;
timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
data += diff * frame_size;
size -= diff * frame_size;
}
diff = stop - cstop;
if (diff > 0) {
offset_end = cstop;
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * frame_size;
}
} else {
gst_buffer_unref (buffer);
return NULL;
}
}
/* Get a metadata writable buffer and apply all changes */
ret = gst_buffer_make_metadata_writable (buffer);
GST_BUFFER_TIMESTAMP (ret) = timestamp;
GST_BUFFER_SIZE (ret) = size;
GST_BUFFER_DATA (ret) = data;
if (change_duration)
GST_BUFFER_DURATION (ret) = duration;
if (change_offset)
GST_BUFFER_OFFSET (ret) = offset;
if (change_offset_end)
GST_BUFFER_OFFSET_END (ret) = offset_end;
return ret;
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioresample *audioresample;
ResampleState *r;
guchar *data, *datacopy;
gulong size;
GstClockTime timestamp;
audioresample = GST_AUDIORESAMPLE (base);
r = audioresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
/* Flush internal samples */
audioresample_pushthrough (audioresample);
/* Inform downstream element about discontinuity */
audioresample->need_discont = TRUE;
/* We want to recalculate the offset */
audioresample->ts_offset = -1;
}
if (audioresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstClockTime stime;
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
audioresample->next_ts = timestamp;
audioresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
audioresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
audioresample->prev_ts = timestamp;
audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
/* need to memdup, resample takes ownership. */
datacopy = g_memdup (data, size);
resample_add_input_data (r, datacopy, size, g_free, datacopy);
return audioresample_do_output (audioresample, outbuf);
}
static GstFlowReturn
gst_file_src_create_read (GstFileSrc * src, guint64 offset, guint length,
GstBuffer ** buffer)
{
int ret;
GstBuffer *buf;
if (G_UNLIKELY (src->read_position != offset)) {
off_t res;
res = lseek (src->fd, offset, SEEK_SET);
if (G_UNLIKELY (res < 0 || res != offset))
goto seek_failed;
src->read_position = offset;
}
buf = gst_buffer_try_new_and_alloc (length);
if (G_UNLIKELY (buf == NULL && length > 0)) {
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", length);
return GST_FLOW_ERROR;
}
/* No need to read anything if length is 0 */
if (length > 0) {
GST_LOG_OBJECT (src, "Reading %d bytes at offset 0x%" G_GINT64_MODIFIER "x",
length, offset);
ret = read (src->fd, GST_BUFFER_DATA (buf), length);
if (G_UNLIKELY (ret < 0))
goto could_not_read;
/* seekable regular files should have given us what we expected */
if (G_UNLIKELY ((guint) ret < length && src->seekable))
goto unexpected_eos;
/* other files should eos if they read 0 and more was requested */
if (G_UNLIKELY (ret == 0 && length > 0))
goto eos;
length = ret;
GST_BUFFER_SIZE (buf) = length;
GST_BUFFER_OFFSET (buf) = offset;
GST_BUFFER_OFFSET_END (buf) = offset + length;
src->read_position += length;
}
*buffer = buf;
return GST_FLOW_OK;
/* ERROR */
seek_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
return GST_FLOW_ERROR;
}
could_not_read:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
unexpected_eos:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("unexpected end of file."));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
eos:
{
GST_DEBUG ("non-regular file hits EOS");
gst_buffer_unref (buf);
return GST_FLOW_UNEXPECTED;
}
}
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 : %llu",
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 : %llu - 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;
}
static GstFlowReturn
gst_fake_src_create (GstBaseSrc * basesrc, guint64 offset, guint length,
GstBuffer ** ret)
{
GstFakeSrc *src;
GstBuffer *buf;
GstClockTime time;
gsize size;
src = GST_FAKE_SRC (basesrc);
buf = gst_fake_src_create_buffer (src, &size);
GST_BUFFER_OFFSET (buf) = offset;
if (src->datarate > 0) {
time = (src->bytes_sent * GST_SECOND) / src->datarate;
GST_BUFFER_DURATION (buf) = size * GST_SECOND / src->datarate;
} else if (gst_base_src_is_live (basesrc)) {
GstClock *clock;
clock = gst_element_get_clock (GST_ELEMENT (src));
if (clock) {
time = gst_clock_get_time (clock);
time -= gst_element_get_base_time (GST_ELEMENT (src));
gst_object_unref (clock);
} else {
/* not an error not to have a clock */
time = GST_CLOCK_TIME_NONE;
}
} else {
time = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_DTS (buf) = time;
GST_BUFFER_PTS (buf) = time;
if (!src->silent) {
gchar dts_str[64], pts_str[64], dur_str[64];
gchar *flag_str;
GST_OBJECT_LOCK (src);
g_free (src->last_message);
if (GST_BUFFER_DTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dts_str, sizeof (dts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DTS (buf)));
} else {
g_strlcpy (dts_str, "none", sizeof (dts_str));
}
if (GST_BUFFER_PTS (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (pts_str, sizeof (pts_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_PTS (buf)));
} else {
g_strlcpy (pts_str, "none", sizeof (pts_str));
}
if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE) {
g_snprintf (dur_str, sizeof (dur_str), "%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
} else {
g_strlcpy (dur_str, "none", sizeof (dur_str));
}
flag_str = gst_buffer_get_flags_string (buf);
src->last_message =
g_strdup_printf ("create ******* (%s:%s) (%u bytes, dts: %s, pts:%s"
", duration: %s, offset: %" G_GINT64_FORMAT ", offset_end: %"
G_GINT64_FORMAT ", flags: %08x %s) %p",
GST_DEBUG_PAD_NAME (GST_BASE_SRC_CAST (src)->srcpad), (guint) size,
dts_str, pts_str, dur_str, GST_BUFFER_OFFSET (buf),
GST_BUFFER_OFFSET_END (buf), GST_MINI_OBJECT_CAST (buf)->flags,
flag_str, buf);
g_free (flag_str);
GST_OBJECT_UNLOCK (src);
g_object_notify_by_pspec ((GObject *) src, pspec_last_message);
}
if (src->signal_handoffs) {
GST_LOG_OBJECT (src, "pre handoff emit");
g_signal_emit (src, gst_fake_src_signals[SIGNAL_HANDOFF], 0, buf,
basesrc->srcpad);
GST_LOG_OBJECT (src, "post handoff emit");
}
src->bytes_sent += size;
*ret = buf;
return GST_FLOW_OK;
}
/* encode the CMML head tag and push the CMML headers
*/
static void
gst_cmml_enc_parse_tag_head (GstCmmlEnc * enc, GstCmmlTagHead * head)
{
GList *headers = NULL;
GList *walk;
guchar *head_string;
GstCaps *caps;
GstBuffer *ident_buf, *preamble_buf, *head_buf;
GstBuffer *buffer;
if (enc->preamble == NULL)
goto flow_unexpected;
GST_INFO_OBJECT (enc, "parsing head tag");
enc->flow_return = gst_cmml_enc_new_ident_header (enc, &ident_buf);
if (enc->flow_return != GST_FLOW_OK)
goto alloc_error;
headers = g_list_append (headers, ident_buf);
enc->flow_return = gst_cmml_enc_new_buffer (enc,
enc->preamble, strlen ((gchar *) enc->preamble), &preamble_buf);
if (enc->flow_return != GST_FLOW_OK)
goto alloc_error;
headers = g_list_append (headers, preamble_buf);
head_string = gst_cmml_parser_tag_head_to_string (enc->parser, head);
enc->flow_return = gst_cmml_enc_new_buffer (enc,
head_string, strlen ((gchar *) head_string), &head_buf);
g_free (head_string);
if (enc->flow_return != GST_FLOW_OK)
goto alloc_error;
headers = g_list_append (headers, head_buf);
caps = gst_pad_get_caps (enc->srcpad);
caps = gst_cmml_enc_set_header_on_caps (enc, caps,
ident_buf, preamble_buf, head_buf);
while (headers) {
buffer = GST_BUFFER (headers->data);
/* set granulepos 0 on headers */
GST_BUFFER_OFFSET_END (buffer) = 0;
gst_buffer_set_caps (buffer, caps);
enc->flow_return = gst_cmml_enc_push (enc, buffer);
headers = g_list_delete_link (headers, headers);
if (GST_FLOW_IS_FATAL (enc->flow_return))
goto push_error;
}
gst_caps_unref (caps);
enc->sent_headers = TRUE;
return;
flow_unexpected:
GST_ELEMENT_ERROR (enc, STREAM, ENCODE,
(NULL), ("got head tag before preamble"));
enc->flow_return = GST_FLOW_ERROR;
return;
push_error:
gst_caps_unref (caps);
/* fallthrough */
alloc_error:
for (walk = headers; walk; walk = walk->next)
gst_buffer_unref (GST_BUFFER (walk->data));
g_list_free (headers);
return;
}
static GstFlowReturn
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
{
GstGSMDec *gsmdec;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
gint needed;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (gsmdec->adapter);
gsmdec->next_ts = GST_CLOCK_TIME_NONE;
/* FIXME, do some good offset */
gsmdec->next_of = 0;
}
gst_adapter_push (gsmdec->adapter, buf);
needed = 33;
/* do we have enough bytes to read a frame */
while (gst_adapter_available (gsmdec->adapter) >= needed) {
GstBuffer *outbuf;
/* always the same amount of output samples */
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
/* If we are not given any timestamp, interpolate from last seen
* timestamp (if any). */
if (timestamp == GST_CLOCK_TIME_NONE)
timestamp = gsmdec->next_ts;
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* interpolate in the next run */
if (timestamp != GST_CLOCK_TIME_NONE)
gsmdec->next_ts = timestamp + gsmdec->duration;
timestamp = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
if (gsmdec->next_of != -1)
gsmdec->next_of += ENCODED_SAMPLES;
GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
/* now encode frame into the output buffer */
data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
if (gsm_decode (gsmdec->state, data,
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
/* invalid frame */
GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
}
gst_adapter_flush (gsmdec->adapter, needed);
/* WAV49 requires alternating 33 and 32 bytes of input */
if (gsmdec->use_wav49)
needed = (needed == 33 ? 32 : 33);
GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
/* push */
ret = gst_pad_push (gsmdec->srcpad, outbuf);
}
gst_object_unref (gsmdec);
return ret;
}
static GstFlowReturn
gst_inter_video_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
{
GstInterVideoSrc *intervideosrc = GST_INTER_VIDEO_SRC (src);
GstBuffer *buffer;
GST_DEBUG_OBJECT (intervideosrc, "create");
buffer = NULL;
g_mutex_lock (intervideosrc->surface->mutex);
if (intervideosrc->surface->video_buffer) {
buffer = gst_buffer_ref (intervideosrc->surface->video_buffer);
intervideosrc->surface->video_buffer_count++;
if (intervideosrc->surface->video_buffer_count >= 30) {
gst_buffer_unref (intervideosrc->surface->video_buffer);
intervideosrc->surface->video_buffer = NULL;
}
}
g_mutex_unlock (intervideosrc->surface->mutex);
if (buffer == NULL) {
GstMapInfo map;
buffer =
gst_buffer_new_and_alloc (GST_VIDEO_INFO_SIZE (&intervideosrc->info));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
memset (map.data, 16, GST_VIDEO_INFO_COMP_STRIDE (&intervideosrc->info, 0) *
GST_VIDEO_INFO_COMP_HEIGHT (&intervideosrc->info, 0));
memset (map.data + GST_VIDEO_INFO_COMP_OFFSET (&intervideosrc->info, 1),
128,
2 * GST_VIDEO_INFO_COMP_STRIDE (&intervideosrc->info, 1) *
GST_VIDEO_INFO_COMP_HEIGHT (&intervideosrc->info, 1));
gst_buffer_unmap (buffer, &map);
}
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (GST_SECOND * intervideosrc->n_frames,
GST_VIDEO_INFO_FPS_D (&intervideosrc->info),
GST_VIDEO_INFO_FPS_N (&intervideosrc->info));
GST_DEBUG_OBJECT (intervideosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND * (intervideosrc->n_frames + 1),
GST_VIDEO_INFO_FPS_D (&intervideosrc->info),
GST_VIDEO_INFO_FPS_N (&intervideosrc->info)) -
GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_OFFSET (buffer) = intervideosrc->n_frames;
GST_BUFFER_OFFSET_END (buffer) = -1;
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (intervideosrc->n_frames == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
}
intervideosrc->n_frames++;
*buf = buffer;
return GST_FLOW_OK;
}
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
GstFlowReturn *flow;
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
srcpad_last_return;
*flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
count, GST_PAD_CAPS (pad), &buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
count, (wid->correction) ? "correction " : "");
gst_wavpack_read_header (&wph, block);
/* Only set when pushing the first buffer again, in that case
* we don't want to delay the buffer or push newsegment events
*/
if (!wid->passthrough) {
/* Only push complete blocks */
if (enc->pending_buffer == NULL) {
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
} else if (enc->pending_offset == wph.block_index) {
enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (enc->pending_buffer);
enc->pending_buffer = buffer;
enc->pending_offset = wph.block_index;
}
if (!(wph.flags & FINAL_BLOCK))
return TRUE;
buffer = enc->pending_buffer;
enc->pending_buffer = NULL;
enc->pending_offset = 0;
/* if it's the first wavpack block, send a NEW_SEGMENT event */
if (wph.block_index == 0) {
gst_pad_push_event (pad,
gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
enc->first_block =
g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
enc->first_block_size = GST_BUFFER_SIZE (buffer);
}
}
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
enc->samplerate);
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
/* push the buffer and forward errors */
GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
GST_BUFFER_SIZE (buffer));
*flow = gst_pad_push (pad, buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
return TRUE;
}
GstFlowReturn
gst_base_video_decoder_have_frame (GstBaseVideoDecoder * base_video_decoder)
{
GstVideoFrame *frame = base_video_decoder->current_frame;
GstBuffer *buffer;
GstBaseVideoDecoderClass *base_video_decoder_class;
GstFlowReturn ret = GST_FLOW_OK;
int n_available;
GST_DEBUG ("have_frame");
base_video_decoder_class =
GST_BASE_VIDEO_DECODER_GET_CLASS (base_video_decoder);
n_available = gst_adapter_available (base_video_decoder->output_adapter);
if (n_available) {
buffer = gst_adapter_take_buffer (base_video_decoder->output_adapter,
n_available);
} else {
buffer = gst_buffer_new_and_alloc (0);
}
frame->distance_from_sync = base_video_decoder->distance_from_sync;
base_video_decoder->distance_from_sync++;
#if 0
if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
frame->presentation_timestamp =
gst_base_video_decoder_get_timestamp (base_video_decoder,
frame->presentation_frame_number);
frame->presentation_duration =
gst_base_video_decoder_get_timestamp (base_video_decoder,
frame->presentation_frame_number + 1) - frame->presentation_timestamp;
frame->decode_timestamp =
gst_base_video_decoder_get_timestamp (base_video_decoder,
frame->decode_frame_number);
}
#endif
#if 0
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;
#endif
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);
}
if (base_video_decoder->discont) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
base_video_decoder->discont = FALSE;
}
frame->sink_buffer = buffer;
base_video_decoder->frames = g_list_append (base_video_decoder->frames,
frame);
/* do something with frame */
ret = base_video_decoder_class->handle_frame (base_video_decoder, frame);
if (!GST_FLOW_IS_SUCCESS (ret)) {
GST_DEBUG ("flow error!");
}
/* create new frame */
base_video_decoder->current_frame =
gst_base_video_decoder_new_frame (base_video_decoder);
return ret;
}
static GstFlowReturn
test_injector_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret;
GstPad *srcpad;
srcpad = gst_element_get_pad (GST_ELEMENT (GST_PAD_PARENT (pad)), "src");
/* since we're increasing timestamp/offsets, push this one first */
GST_LOG (" passing buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
"], offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
gst_buffer_ref (buf);
ret = gst_pad_push (srcpad, buf);
if (g_random_double () < injector_inject_probability) {
GstBuffer *ibuf;
ibuf = gst_buffer_copy (buf);
if (GST_BUFFER_OFFSET_IS_VALID (buf) &&
GST_BUFFER_OFFSET_END_IS_VALID (buf)) {
guint64 delta;
delta = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
GST_BUFFER_OFFSET (ibuf) += delta / 4;
GST_BUFFER_OFFSET_END (ibuf) += delta / 4;
} else {
GST_BUFFER_OFFSET (ibuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (ibuf) = GST_BUFFER_OFFSET_NONE;
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) &&
GST_BUFFER_DURATION_IS_VALID (buf)) {
GstClockTime delta;
delta = GST_BUFFER_DURATION (buf);
GST_BUFFER_TIMESTAMP (ibuf) += delta / 4;
} else {
GST_BUFFER_TIMESTAMP (ibuf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (ibuf) = GST_CLOCK_TIME_NONE;
}
if (GST_BUFFER_TIMESTAMP_IS_VALID (ibuf) ||
GST_BUFFER_OFFSET_IS_VALID (ibuf)) {
GST_LOG ("injecting buffer [t=%" GST_TIME_FORMAT "-%" GST_TIME_FORMAT
"], offset=%" G_GINT64_FORMAT ", offset_end=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (ibuf) +
GST_BUFFER_DURATION (ibuf)), GST_BUFFER_OFFSET (ibuf),
GST_BUFFER_OFFSET_END (ibuf));
if (gst_pad_push (srcpad, ibuf) != GST_FLOW_OK) {
/* ignore return value */
}
} else {
GST_WARNING ("couldn't inject buffer, no incoming timestamps or offsets");
gst_buffer_unref (ibuf);
}
}
gst_buffer_unref (buf);
return ret;
}
static GstFlowReturn
gst_base_video_decoder_chain (GstPad * pad, GstBuffer * buf)
{
GstBaseVideoDecoder *base_video_decoder;
GstBaseVideoDecoderClass *klass;
GstBuffer *buffer;
GstFlowReturn ret;
GST_DEBUG ("chain %" G_GINT64_FORMAT, GST_BUFFER_TIMESTAMP (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));
klass = GST_BASE_VIDEO_DECODER_GET_CLASS (base_video_decoder);
GST_DEBUG_OBJECT (base_video_decoder, "chain");
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (base_video_decoder, "received DISCONT buffer");
if (base_video_decoder->started) {
gst_base_video_decoder_reset (base_video_decoder);
}
}
if (!base_video_decoder->started) {
klass->start (base_video_decoder);
base_video_decoder->started = TRUE;
}
if (GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
GST_DEBUG ("timestamp %" G_GINT64_FORMAT " offset %" G_GINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buf), base_video_decoder->offset);
base_video_decoder->last_sink_timestamp = GST_BUFFER_TIMESTAMP (buf);
}
if (GST_BUFFER_OFFSET_END (buf) != -1) {
GST_DEBUG ("gp %" G_GINT64_FORMAT, GST_BUFFER_OFFSET_END (buf));
base_video_decoder->last_sink_offset_end = GST_BUFFER_OFFSET_END (buf);
}
base_video_decoder->offset += GST_BUFFER_SIZE (buf);
#if 0
if (base_video_decoder->timestamp_offset == GST_CLOCK_TIME_NONE &&
GST_BUFFER_TIMESTAMP (buf) != GST_CLOCK_TIME_NONE) {
GST_DEBUG ("got new offset %lld", GST_BUFFER_TIMESTAMP (buf));
base_video_decoder->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
}
#endif
if (base_video_decoder->current_frame == NULL) {
base_video_decoder->current_frame =
gst_base_video_decoder_new_frame (base_video_decoder);
}
gst_adapter_push (base_video_decoder->input_adapter, buf);
if (!base_video_decoder->have_sync) {
int n, m;
GST_DEBUG ("no sync, scanning");
n = gst_adapter_available (base_video_decoder->input_adapter);
m = klass->scan_for_sync (base_video_decoder, FALSE, 0, n);
if (m < 0) {
g_warning ("subclass returned negative scan %d", m);
}
if (m >= n) {
g_warning ("subclass scanned past end %d >= %d", m, n);
}
gst_adapter_flush (base_video_decoder->input_adapter, m);
if (m < n) {
GST_DEBUG ("found possible sync after %d bytes (of %d)", m, n);
/* this is only "maybe" sync */
base_video_decoder->have_sync = TRUE;
}
if (!base_video_decoder->have_sync) {
gst_object_unref (base_video_decoder);
return GST_FLOW_OK;
}
}
/* FIXME: use gst_adapter_prev_timestamp() here instead? */
buffer = gst_adapter_get_buffer (base_video_decoder->input_adapter);
base_video_decoder->buffer_timestamp = GST_BUFFER_TIMESTAMP (buffer);
gst_buffer_unref (buffer);
do {
ret = klass->parse_data (base_video_decoder, FALSE);
} while (ret == GST_FLOW_OK);
if (ret == GST_BASE_VIDEO_DECODER_FLOW_NEED_DATA) {
gst_object_unref (base_video_decoder);
return GST_FLOW_OK;
}
gst_object_unref (base_video_decoder);
return ret;
}
/* this tests that the output is a correct discontinuous stream
* if the input is; ie input drops in time come out the same way */
static void
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *audioresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
int i, j;
GstMapInfo map;
gint16 *p;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
audioresample =
setup_audioresample (2, 3, inrate, outrate, GST_AUDIO_NE (S16));
caps = gst_pad_get_current_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
/* "drop" half the buffers */
ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
GST_BUFFER_TIMESTAMP (inbuffer) = ints;
GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
gst_buffer_map (inbuffer, &map, GST_MAP_WRITE);
p = (gint16 *) map.data;
/* create a 16 bit signed ramp */
for (i = 0; i < samples; ++i) {
*p = -32767 + i * (65535 / samples);
++p;
*p = -32767 + i * (65535 / samples);
++p;
}
gst_buffer_unmap (inbuffer, &map);
GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* check if the timestamp of the pushed buffer matches the incoming one */
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
fail_if (outbuffer == NULL);
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
if (j > 1) {
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
"expected discont for buffer #%d", j);
}
}
/* cleanup */
gst_caps_unref (caps);
cleanup_audioresample (audioresample);
}
GstFlowReturn
gst_base_video_decoder_finish_frame (GstBaseVideoDecoder * base_video_decoder,
GstVideoFrame * frame)
{
GstBaseVideoDecoderClass *base_video_decoder_class;
GstBuffer *src_buffer;
GST_DEBUG ("finish frame");
base_video_decoder_class =
GST_BASE_VIDEO_DECODER_GET_CLASS (base_video_decoder);
GST_DEBUG ("finish frame sync=%d pts=%" G_GINT64_FORMAT, frame->is_sync_point,
frame->presentation_timestamp);
if (frame->is_sync_point) {
if (GST_CLOCK_TIME_IS_VALID (frame->presentation_timestamp)) {
if (frame->presentation_timestamp != base_video_decoder->timestamp_offset) {
GST_DEBUG ("sync timestamp %" G_GINT64_FORMAT " diff %" G_GINT64_FORMAT,
frame->presentation_timestamp,
frame->presentation_timestamp -
base_video_decoder->state.segment.start);
base_video_decoder->timestamp_offset = frame->presentation_timestamp;
base_video_decoder->field_index = 0;
} else {
/* This case is for one initial timestamp and no others, e.g.,
* filesrc ! decoder ! xvimagesink */
GST_WARNING ("sync timestamp didn't change, ignoring");
frame->presentation_timestamp = GST_CLOCK_TIME_NONE;
}
} else {
GST_WARNING ("sync point doesn't have timestamp");
if (GST_CLOCK_TIME_IS_VALID (base_video_decoder->timestamp_offset)) {
GST_ERROR ("No base timestamp. Assuming frames start at 0");
base_video_decoder->timestamp_offset = 0;
base_video_decoder->field_index = 0;
}
}
}
frame->field_index = base_video_decoder->field_index;
base_video_decoder->field_index += frame->n_fields;
if (frame->presentation_timestamp == GST_CLOCK_TIME_NONE) {
frame->presentation_timestamp =
gst_base_video_decoder_get_field_timestamp (base_video_decoder,
frame->field_index);
frame->presentation_duration = GST_CLOCK_TIME_NONE;
frame->decode_timestamp =
gst_base_video_decoder_get_timestamp (base_video_decoder,
frame->decode_frame_number);
}
if (frame->presentation_duration == GST_CLOCK_TIME_NONE) {
frame->presentation_duration =
gst_base_video_decoder_get_field_timestamp (base_video_decoder,
frame->field_index + frame->n_fields) - frame->presentation_timestamp;
}
if (GST_CLOCK_TIME_IS_VALID (base_video_decoder->last_timestamp)) {
if (frame->presentation_timestamp < base_video_decoder->last_timestamp) {
GST_WARNING ("decreasing timestamp (%" G_GINT64_FORMAT " < %"
G_GINT64_FORMAT ")", frame->presentation_timestamp,
base_video_decoder->last_timestamp);
}
}
base_video_decoder->last_timestamp = frame->presentation_timestamp;
GST_BUFFER_FLAG_UNSET (frame->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
int tff = base_video_decoder->state.top_field_first;
if (frame->field_index & 1) {
tff ^= 1;
}
if (tff) {
GST_BUFFER_FLAG_SET (frame->src_buffer, GST_VIDEO_BUFFER_TFF);
} else {
GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_VIDEO_BUFFER_TFF);
}
GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_VIDEO_BUFFER_RFF);
GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
if (frame->n_fields == 3) {
GST_BUFFER_FLAG_SET (frame->src_buffer, GST_VIDEO_BUFFER_RFF);
} else if (frame->n_fields == 1) {
GST_BUFFER_FLAG_UNSET (frame->src_buffer, GST_VIDEO_BUFFER_ONEFIELD);
}
}
GST_BUFFER_TIMESTAMP (frame->src_buffer) = frame->presentation_timestamp;
GST_BUFFER_DURATION (frame->src_buffer) = frame->presentation_duration;
GST_BUFFER_OFFSET (frame->src_buffer) = -1;
GST_BUFFER_OFFSET_END (frame->src_buffer) = -1;
GST_DEBUG ("pushing frame %" G_GINT64_FORMAT, frame->presentation_timestamp);
base_video_decoder->frames =
g_list_remove (base_video_decoder->frames, frame);
gst_base_video_decoder_set_src_caps (base_video_decoder);
src_buffer = frame->src_buffer;
frame->src_buffer = NULL;
gst_base_video_decoder_free_frame (frame);
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->state.segment, GST_FORMAT_TIME,
start, stop, &start, &stop)) {
GST_BUFFER_TIMESTAMP (src_buffer) = start;
GST_BUFFER_DURATION (src_buffer) = stop - start;
} else {
GST_DEBUG ("dropping buffer outside segment");
gst_buffer_unref (src_buffer);
return GST_FLOW_OK;
}
}
return gst_pad_push (GST_BASE_VIDEO_CODEC_SRC_PAD (base_video_decoder),
src_buffer);
}
static GstFlowReturn
gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
GstClockTime timestamp, expected_timestamp;
gint channels = GST_AUDIO_FILTER_CHANNELS (self);
gint rate = GST_AUDIO_FILTER_RATE (self);
gint bps = GST_AUDIO_FILTER_BPS (self);
GstMapInfo inmap, outmap;
guint input_samples;
guint output_samples;
guint generated_samples;
guint64 output_offset;
gint64 diff = 0;
GstClockTime stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (!GST_CLOCK_TIME_IS_VALID (timestamp)
&& !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
GST_ERROR_OBJECT (self, "Invalid timestamp");
return GST_FLOW_ERROR;
}
g_mutex_lock (&self->lock);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (self), stream_time);
g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
if (GST_CLOCK_TIME_IS_VALID (self->start_ts))
expected_timestamp =
self->start_ts + gst_util_uint64_scale_int (self->nsamples_in,
GST_SECOND, rate);
else
expected_timestamp = GST_CLOCK_TIME_NONE;
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf)
|| (GST_CLOCK_TIME_IS_VALID (expected_timestamp)
&& (ABS (GST_CLOCK_DIFF (timestamp,
expected_timestamp) > 5 * GST_MSECOND)))) {
GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
if (GST_CLOCK_TIME_IS_VALID (expected_timestamp))
gst_audio_fx_base_fir_filter_push_residue (self);
self->buffer_fill = 0;
g_free (self->buffer);
self->buffer = NULL;
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
self->nsamples_out = 0;
self->nsamples_in = 0;
} else if (!GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
self->start_ts = timestamp;
self->start_off = GST_BUFFER_OFFSET (inbuf);
}
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
input_samples = (inmap.size / bps) / channels;
output_samples = (outmap.size / bps) / channels;
self->nsamples_in += input_samples;
generated_samples =
self->process (self, inmap.data, outmap.data, input_samples);
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
g_assert (generated_samples <= output_samples);
self->nsamples_out += generated_samples;
if (generated_samples == 0)
goto no_samples;
/* Calculate the number of samples we can push out now without outputting
* latency zeros in the beginning */
diff = ((gint64) self->nsamples_out) - ((gint64) self->latency);
if (diff < 0)
goto no_samples;
if (diff < generated_samples) {
gint64 tmp = diff;
diff = generated_samples - diff;
generated_samples = tmp;
} else {
diff = 0;
}
gst_buffer_resize (outbuf, diff * bps * channels,
generated_samples * bps * channels);
output_offset = self->nsamples_out - self->latency - generated_samples;
GST_BUFFER_TIMESTAMP (outbuf) =
self->start_ts + gst_util_uint64_scale_int (output_offset, GST_SECOND,
rate);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (output_samples, GST_SECOND, rate);
if (self->start_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->start_off + output_offset;
GST_BUFFER_OFFSET_END (outbuf) =
GST_BUFFER_OFFSET (outbuf) + generated_samples;
} else {
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
}
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (self,
"Pushing buffer of size %" G_GSIZE_FORMAT " with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
gst_buffer_get_size (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), generated_samples);
return GST_FLOW_OK;
no_samples:
{
g_mutex_unlock (&self->lock);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
}
}
static GstFlowReturn
audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
{
int outsize;
int outsamples;
ResampleState *r;
r = audioresample->resample;
outsize = resample_get_output_size (r);
GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
/* protect against mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
GST_WARNING_OBJECT (audioresample,
"overriding audioresample's outsize %d with outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
outsize = GST_BUFFER_SIZE (outbuf);
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
GST_WARNING_OBJECT (audioresample,
"audioresample's outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
outsamples = outsize / r->sample_size;
GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
if (audioresample->ts_offset != -1) {
audioresample->offset += outsamples;
audioresample->ts_offset += outsamples;
audioresample->next_ts =
gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
audioresample->o_rate);
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND,
audioresample->o_rate);
}
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
/* this is an error that when it happens, would need fixing in the
* resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
* and it gave us more ! */
GST_WARNING_OBJECT (audioresample,
"audioresample, you memory corrupting bastard. "
"you gave me outsize %d while my buffer was size %d",
outsize, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_ERROR;
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
GST_WARNING_OBJECT (audioresample,
"audioresample's written outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
GST_BUFFER_SIZE (outbuf) = outsize;
if (G_UNLIKELY (audioresample->need_discont)) {
GST_DEBUG_OBJECT (audioresample,
"marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
audioresample->need_discont = FALSE;
}
GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
return GST_FLOW_OK;
}