static GstFlowReturn
vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
{
GstFlowReturn result;
/* clip */
if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
dec->vi.channels * dec->width))) {
GST_LOG_OBJECT (dec, "clipped buffer");
return GST_FLOW_OK;
}
if (dec->discont) {
GST_LOG_OBJECT (dec, "setting DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
dec->discont = FALSE;
}
GST_DEBUG_OBJECT (dec,
"pushing time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
result = gst_pad_push (dec->srcpad, buf);
return result;
}
static GstFlowReturn
gst_a52dec_push (GstA52Dec * a52dec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
GstFlowReturn result;
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
result =
gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
if (result != GST_FLOW_OK)
return result;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
result = GST_FLOW_OK;
if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
/* set discont when needed */
if (a52dec->discont) {
GST_LOG_OBJECT (a52dec, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
a52dec->discont = FALSE;
}
if (a52dec->segment.rate > 0.0) {
GST_DEBUG_OBJECT (a52dec,
"Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
result = gst_pad_push (srcpad, buf);
} else {
/* reverse playback, queue frame till later when we get a discont. */
GST_DEBUG_OBJECT (a52dec, "queued frame");
a52dec->queued = g_list_prepend (a52dec->queued, buf);
}
}
return result;
}
static GstBuffer *
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
GST_OBJECT_LOCK (bpad);
buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return buffer;
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
goto invalid_header;
if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
if (!(wph.flags & INITIAL_BLOCK))
goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf);
dec->wv_id.length = GST_BUFFER_SIZE (buf);
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
if (!dec->context) {
GST_WARNING ("Couldn't decode buffer: %s", error_msg);
dec->error_count++;
if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
goto out; /* just return OK for now */
} else {
goto decode_error;
}
}
}
g_assert (dec->context != NULL);
dec->error_count = 0;
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBitsPerSample (dec->context)) ||
#ifdef WAVPACK_OLD_API
(dec->channel_mask != dec->context->config.channel_mask);
#else
(dec->channel_mask != WavpackGetChannelMask (dec->context));
#endif
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
gint channel_mask;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
#ifdef WAVPACK_OLD_API
channel_mask = dec->context->config.channel_mask;
#else
channel_mask = WavpackGetChannelMask (dec->context);
#endif
if (channel_mask == 0)
channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
dec->channel_mask = channel_mask;
/* Only set the channel layout for more than two channels
* otherwise things break unfortunately */
if (channel_mask != 0 && dec->channels > 2)
if (!gst_wavpack_set_channel_layout (caps, channel_mask))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
/* If we got a DISCONT buffer forward the flag. Nothing else
* has to be done as libwavpack doesn't store state between
* Wavpack blocks */
if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
dec->next_block_index = wph.block_index + wph.block_samples;
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
dec->sample_rate, 4 * dec->channels))) {
GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (dec->srcpad, outbuf);
}
out:
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
gst_buffer_unref (buf);
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
decode_error:
{
const gchar *reason = "unknown";
if (dec->context) {
#ifdef WAVPACK_OLD_API
reason = dec->context->error_message;
#else
reason = WavpackGetErrorMessage (dec->context);
#endif
} else {
reason = "couldn't create decoder context";
}
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream: %s", reason));
gst_buffer_unref (outbuf);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_audio_segment_clip_clip_buffer (GstSegmentClip * base, GstBuffer * buffer,
GstBuffer ** outbuf)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
GstSegment *segment = &base->segment;
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
GstClockTime duration = GST_BUFFER_DURATION (buffer);
guint64 offset = GST_BUFFER_OFFSET (buffer);
guint64 offset_end = GST_BUFFER_OFFSET_END (buffer);
guint size = gst_buffer_get_size (buffer);
if (!self->rate || !self->framesize) {
GST_ERROR_OBJECT (self, "Not negotiated yet");
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
if (segment->format != GST_FORMAT_DEFAULT &&
segment->format != GST_FORMAT_TIME) {
GST_DEBUG_OBJECT (self, "Unsupported segment format %s",
gst_format_get_name (segment->format));
*outbuf = buffer;
return GST_FLOW_OK;
}
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
GST_WARNING_OBJECT (self, "Buffer without valid timestamp");
*outbuf = buffer;
return GST_FLOW_OK;
}
*outbuf =
gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize);
if (!*outbuf) {
GST_DEBUG_OBJECT (self, "Buffer outside the configured segment");
/* Now return unexpected if we're before/after the end */
if (segment->format == GST_FORMAT_TIME) {
if (segment->rate >= 0) {
if (segment->stop != -1 && timestamp >= segment->stop)
return GST_FLOW_EOS;
} else {
if (!GST_CLOCK_TIME_IS_VALID (duration))
duration =
gst_util_uint64_scale_int (size, GST_SECOND,
self->framesize * self->rate);
if (segment->start != -1 && timestamp + duration <= segment->start)
return GST_FLOW_EOS;
}
} else {
if (segment->rate >= 0) {
if (segment->stop != -1 && offset != -1 && offset >= segment->stop)
return GST_FLOW_EOS;
} else if (offset != -1 || offset_end != -1) {
if (offset_end == -1)
offset_end = offset + size / self->framesize;
if (segment->start != -1 && offset_end <= segment->start)
return GST_FLOW_EOS;
}
}
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_timecodewait_asink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstClockTime timestamp;
GstTimeCodeWait *self = GST_TIMECODEWAIT (parent);
GstClockTime current_running_time;
GstClockTime video_running_time = GST_CLOCK_TIME_NONE;
GstClockTime duration;
GstClockTime running_time_at_end = GST_CLOCK_TIME_NONE;
gint asign, vsign = 1, esign = 1;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
if (timestamp == GST_CLOCK_TIME_NONE) {
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
g_mutex_lock (&self->mutex);
self->asegment.position = timestamp;
asign =
gst_segment_to_running_time_full (&self->asegment, GST_FORMAT_TIME,
self->asegment.position, ¤t_running_time);
if (asign == 0) {
g_mutex_unlock (&self->mutex);
gst_buffer_unref (inbuf);
GST_ERROR_OBJECT (self, "Could not get current running time");
return GST_FLOW_ERROR;
}
if (self->vsegment.format == GST_FORMAT_TIME) {
vsign =
gst_segment_to_running_time_full (&self->vsegment, GST_FORMAT_TIME,
self->vsegment.position, &video_running_time);
if (vsign == 0) {
video_running_time = GST_CLOCK_TIME_NONE;
}
}
while (!(self->video_eos_flag || self->audio_flush_flag
|| self->shutdown_flag) && (video_running_time == GST_CLOCK_TIME_NONE
|| gst_timecodewait_compare_guint64_with_signs (asign,
current_running_time, vsign, video_running_time) == 1)
&& self->running_time_of_timecode == GST_CLOCK_TIME_NONE) {
g_cond_wait (&self->cond, &self->mutex);
vsign =
gst_segment_to_running_time_full (&self->vsegment, GST_FORMAT_TIME,
self->vsegment.position, &video_running_time);
if (vsign == 0) {
video_running_time = GST_CLOCK_TIME_NONE;
}
}
if (self->audio_flush_flag || self->shutdown_flag) {
GST_DEBUG_OBJECT (self, "Shutting down, ignoring frame");
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
}
duration = GST_BUFFER_DURATION (inbuf);
if (duration != GST_CLOCK_TIME_NONE) {
esign =
gst_segment_to_running_time_full (&self->asegment, GST_FORMAT_TIME,
self->asegment.position + duration, &running_time_at_end);
if (esign == 0) {
g_mutex_unlock (&self->mutex);
GST_ERROR_OBJECT (self, "Could not get running time at end");
gst_buffer_unref (inbuf);
return GST_FLOW_ERROR;
}
}
if (self->running_time_of_timecode == GST_CLOCK_TIME_NONE
|| gst_timecodewait_compare_guint64_with_signs (esign,
running_time_at_end, 1, self->running_time_of_timecode) == -1) {
GST_DEBUG_OBJECT (self,
"Dropped an audio buf at %" GST_TIME_FORMAT " with timecode %"
GST_TIME_FORMAT " video timecode %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_running_time),
GST_TIME_ARGS (self->running_time_of_timecode),
GST_TIME_ARGS (video_running_time));
gst_buffer_unref (inbuf);
inbuf = NULL;
} else if (current_running_time < self->running_time_of_timecode
&& running_time_at_end > self->running_time_of_timecode) {
GstSegment asegment2 = self->asegment;
gst_segment_set_running_time (&asegment2, GST_FORMAT_TIME,
self->running_time_of_timecode);
inbuf =
gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate,
self->ainfo.bpf);
}
g_mutex_unlock (&self->mutex);
if (inbuf)
return gst_pad_push (self->asrcpad, inbuf);
else
return GST_FLOW_OK;
}
void test_buffer_clipping_samples()
{
GstSegment s;
GstBuffer *buf;
GstBuffer *ret;
guint8 *data;
xmlfile = "test_buffer_clipping_samples";
std_log(LOG_FILENAME_LINE, "Test Started test_buffer_clipping_samples");
/* Clip start and end */
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 400,
800, 400);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buf) = 10 * GST_SECOND;
GST_BUFFER_OFFSET (buf) = 200;
GST_BUFFER_OFFSET_END (buf) = 1200;
ret = gst_audio_buffer_clip (buf, &s, 100, 1);
fail_unless (ret != NULL);
fail_unless (GST_BUFFER_TIMESTAMP (ret) == 4 * GST_SECOND);
fail_unless (GST_BUFFER_DURATION (ret) == 4 * GST_SECOND);
fail_unless (GST_BUFFER_OFFSET (ret) == 400);
fail_unless (GST_BUFFER_OFFSET_END (ret) == 800);
fail_unless (GST_BUFFER_DATA (ret) == data + 200);
fail_unless (GST_BUFFER_SIZE (ret) == 400);
gst_buffer_unref (ret);
/* Clip only start */
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 400,
1200, 400);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buf) = 10 * GST_SECOND;
GST_BUFFER_OFFSET (buf) = 200;
GST_BUFFER_OFFSET_END (buf) = 1200;
ret = gst_audio_buffer_clip (buf, &s, 100, 1);
fail_unless (ret != NULL);
fail_unless (GST_BUFFER_TIMESTAMP (ret) == 4 * GST_SECOND);
fail_unless (GST_BUFFER_DURATION (ret) == 8 * GST_SECOND);
fail_unless (GST_BUFFER_OFFSET (ret) == 400);
fail_unless (GST_BUFFER_OFFSET_END (ret) == 1200);
fail_unless (GST_BUFFER_DATA (ret) == data + 200);
fail_unless (GST_BUFFER_SIZE (ret) == 800);
gst_buffer_unref (ret);
/* Clip only stop */
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 200,
1000, 200);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buf) = 10 * GST_SECOND;
GST_BUFFER_OFFSET (buf) = 200;
GST_BUFFER_OFFSET_END (buf) = 1200;
ret = gst_audio_buffer_clip (buf, &s, 100, 1);
fail_unless (ret != NULL);
fail_unless (GST_BUFFER_TIMESTAMP (ret) == 2 * GST_SECOND);
fail_unless (GST_BUFFER_DURATION (ret) == 8 * GST_SECOND);
fail_unless (GST_BUFFER_OFFSET (ret) == 200);
fail_unless (GST_BUFFER_OFFSET_END (ret) == 1000);
fail_unless (GST_BUFFER_DATA (ret) == data);
fail_unless (GST_BUFFER_SIZE (ret) == 800);
gst_buffer_unref (ret);
/* Buffer outside segment */
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 1200,
2000, 1200);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buf) = 10 * GST_SECOND;
GST_BUFFER_OFFSET (buf) = 200;
GST_BUFFER_OFFSET_END (buf) = 1200;
ret = gst_audio_buffer_clip (buf, &s, 100, 1);
fail_unless (ret == NULL);
/* Clip start and end but don't touch duration and offset_end */
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 400,
800, 400);
GST_BUFFER_TIMESTAMP (buf) = 2 * GST_SECOND;
GST_BUFFER_DURATION (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET (buf) = 200;
GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET_NONE;
ret = gst_audio_buffer_clip (buf, &s, 100, 1);
fail_unless (ret != NULL);
fail_unless (GST_BUFFER_TIMESTAMP (ret) == 4 * GST_SECOND);
fail_unless (GST_BUFFER_DURATION (ret) == GST_CLOCK_TIME_NONE);
fail_unless (GST_BUFFER_OFFSET (ret) == 400);
fail_unless (GST_BUFFER_OFFSET_END (ret) == GST_BUFFER_OFFSET_NONE);
fail_unless (GST_BUFFER_DATA (ret) == data + 200);
fail_unless (GST_BUFFER_SIZE (ret) == 400);
gst_buffer_unref (ret);
/* If the buffer has no offset it should assert()
* FIXME: check if return value is the same as the input buffer.
* probably can't be done because the assert() does a SIGABRT.
*/
buf = gst_buffer_new ();
data = (guint8 *) g_malloc (1000);
GST_BUFFER_SIZE (buf) = 1000;
GST_BUFFER_DATA (buf) = GST_BUFFER_MALLOCDATA (buf) = data;
gst_segment_init (&s, GST_FORMAT_DEFAULT);
gst_segment_set_newsegment (&s, FALSE, 1.0, GST_FORMAT_DEFAULT, 0, 10, 0);
GST_BUFFER_TIMESTAMP (buf) = 0 * GST_SECOND;
GST_BUFFER_DURATION (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET (buf) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET_NONE;
ASSERT_CRITICAL (ret = gst_audio_buffer_clip (buf, &s, 100, 1));
gst_buffer_unref (buf);
std_log(LOG_FILENAME_LINE, "Test Successful");
create_xml(0);
}
static GstFlowReturn
gst_live_live_adder_chain (GstPad * pad, GstBuffer * buffer)
{
GstLiveAdder *adder = GST_LIVE_ADDER (gst_pad_get_parent_element (pad));
GstLiveAdderPadPrivate *padprivate = NULL;
GstFlowReturn ret = GST_FLOW_OK;
GList *item = NULL;
GstClockTime skip = 0;
gint64 drift = 0; /* Positive if new buffer after old buffer */
GST_OBJECT_LOCK (adder);
ret = adder->srcresult;
GST_DEBUG ("Incoming buffer time:%" GST_TIME_FORMAT " duration:%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (adder, "Passing non-ok result from src: %s",
gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto out;
}
padprivate = gst_pad_get_element_private (pad);
if (!padprivate) {
ret = GST_FLOW_NOT_LINKED;
gst_buffer_unref (buffer);
goto out;
}
if (padprivate->eos) {
GST_DEBUG_OBJECT (adder, "Received buffer after EOS");
ret = GST_FLOW_UNEXPECTED;
gst_buffer_unref (buffer);
goto out;
}
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
goto invalid_timestamp;
if (padprivate->segment.format == GST_FORMAT_UNDEFINED) {
GST_WARNING_OBJECT (adder, "No new-segment received,"
" initializing segment with time 0..-1");
gst_segment_init (&padprivate->segment, GST_FORMAT_TIME);
gst_segment_set_newsegment (&padprivate->segment,
FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
}
if (padprivate->segment.format != GST_FORMAT_TIME)
goto invalid_segment;
buffer = gst_buffer_make_metadata_writable (buffer);
drift = GST_BUFFER_TIMESTAMP (buffer) - padprivate->expected_timestamp;
/* Just see if we receive invalid timestamp/durations */
if (GST_CLOCK_TIME_IS_VALID (padprivate->expected_timestamp) &&
!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT) &&
(drift != 0)) {
GST_LOG_OBJECT (adder,
"Timestamp discontinuity without the DISCONT flag set"
" (expected %" GST_TIME_FORMAT ", got %" GST_TIME_FORMAT
" drift:%" G_GINT64_FORMAT "ms)",
GST_TIME_ARGS (padprivate->expected_timestamp),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), drift / GST_MSECOND);
/* We accept drifts of 10ms */
if (ABS (drift) < (10 * GST_MSECOND)) {
GST_DEBUG ("Correcting minor drift");
GST_BUFFER_TIMESTAMP (buffer) = padprivate->expected_timestamp;
}
}
/* If there is no duration, lets set one */
if (!GST_BUFFER_DURATION_IS_VALID (buffer)) {
GST_BUFFER_DURATION (buffer) =
gst_audio_duration_from_pad_buffer (pad, buffer);
padprivate->expected_timestamp = GST_CLOCK_TIME_NONE;
} else {
padprivate->expected_timestamp = GST_BUFFER_TIMESTAMP (buffer) +
GST_BUFFER_DURATION (buffer);
}
/*
* Lets clip the buffer to the segment (so we don't have to worry about
* cliping afterwards).
* This should also guarantee us that we'll have valid timestamps and
* durations afterwards
*/
buffer = gst_audio_buffer_clip (buffer, &padprivate->segment, adder->rate,
adder->bps);
/* buffer can be NULL if it's completely outside of the segment */
if (!buffer) {
GST_DEBUG ("Buffer completely outside of configured segment, dropping it");
goto out;
}
/*
* Make sure all incoming buffers share the same timestamping
*/
GST_BUFFER_TIMESTAMP (buffer) =
gst_segment_to_running_time (&padprivate->segment,
padprivate->segment.format, GST_BUFFER_TIMESTAMP (buffer));
if (GST_CLOCK_TIME_IS_VALID (adder->next_timestamp) &&
GST_BUFFER_TIMESTAMP (buffer) < adder->next_timestamp) {
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <
adder->next_timestamp) {
GST_DEBUG_OBJECT (adder, "Buffer is late, dropping (ts: %" GST_TIME_FORMAT
" duration: %" GST_TIME_FORMAT ")",
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
gst_buffer_unref (buffer);
goto out;
} else {
skip = adder->next_timestamp - GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (adder, "Buffer is partially late, skipping %"
GST_TIME_FORMAT, GST_TIME_ARGS (skip));
}
}
/* If our new buffer's head is higher than the queue's head, lets wake up,
* we may not have to wait for as long
*/
if (adder->clock_id &&
g_queue_peek_head (adder->buffers) != NULL &&
GST_BUFFER_TIMESTAMP (buffer) + skip <
GST_BUFFER_TIMESTAMP (g_queue_peek_head (adder->buffers)))
gst_clock_id_unschedule (adder->clock_id);
for (item = g_queue_peek_head_link (adder->buffers);
item; item = g_list_next (item)) {
GstBuffer *oldbuffer = item->data;
GstClockTime old_skip = 0;
GstClockTime mix_duration = 0;
GstClockTime mix_start = 0;
GstClockTime mix_end = 0;
/* We haven't reached our place yet */
if (GST_BUFFER_TIMESTAMP (buffer) + skip >=
GST_BUFFER_TIMESTAMP (oldbuffer) + GST_BUFFER_DURATION (oldbuffer))
continue;
/* We're past our place, lets insert ouselves here */
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <=
GST_BUFFER_TIMESTAMP (oldbuffer))
break;
/* if we reach this spot, we have overlap, so we must mix */
/* First make a subbuffer with the non-overlapping part */
if (GST_BUFFER_TIMESTAMP (buffer) + skip < GST_BUFFER_TIMESTAMP (oldbuffer)) {
GstBuffer *subbuffer = NULL;
GstClockTime subbuffer_duration = GST_BUFFER_TIMESTAMP (oldbuffer) -
(GST_BUFFER_TIMESTAMP (buffer) + skip);
subbuffer = gst_buffer_create_sub (buffer,
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, subbuffer_duration));
GST_BUFFER_TIMESTAMP (subbuffer) = GST_BUFFER_TIMESTAMP (buffer) + skip;
GST_BUFFER_DURATION (subbuffer) = subbuffer_duration;
skip += subbuffer_duration;
g_queue_insert_before (adder->buffers, item, subbuffer);
}
/* Now we are on the overlapping part */
oldbuffer = gst_buffer_make_writable (oldbuffer);
item->data = oldbuffer;
old_skip = GST_BUFFER_TIMESTAMP (buffer) + skip -
GST_BUFFER_TIMESTAMP (oldbuffer);
mix_start = GST_BUFFER_TIMESTAMP (oldbuffer) + old_skip;
if (GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer) <
GST_BUFFER_TIMESTAMP (oldbuffer) + GST_BUFFER_DURATION (oldbuffer))
mix_end = GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
else
mix_end = GST_BUFFER_TIMESTAMP (oldbuffer) +
GST_BUFFER_DURATION (oldbuffer);
mix_duration = mix_end - mix_start;
adder->func (GST_BUFFER_DATA (oldbuffer) +
gst_live_adder_length_from_duration (adder, old_skip),
GST_BUFFER_DATA (buffer) +
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, mix_duration));
skip += mix_duration;
}
g_cond_broadcast (adder->not_empty_cond);
if (skip == GST_BUFFER_DURATION (buffer)) {
gst_buffer_unref (buffer);
} else {
if (skip) {
GstClockTime subbuffer_duration = GST_BUFFER_DURATION (buffer) - skip;
GstClockTime subbuffer_ts = GST_BUFFER_TIMESTAMP (buffer) + skip;
GstBuffer *new_buffer = gst_buffer_create_sub (buffer,
gst_live_adder_length_from_duration (adder, skip),
gst_live_adder_length_from_duration (adder, subbuffer_duration));
gst_buffer_unref (buffer);
buffer = new_buffer;
GST_BUFFER_TIMESTAMP (buffer) = subbuffer_ts;
GST_BUFFER_DURATION (buffer) = subbuffer_duration;
}
if (item)
g_queue_insert_before (adder->buffers, item, buffer);
else
g_queue_push_tail (adder->buffers, buffer);
}
out:
GST_OBJECT_UNLOCK (adder);
gst_object_unref (adder);
return ret;
invalid_timestamp:
GST_OBJECT_UNLOCK (adder);
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (adder, STREAM, FAILED,
("Buffer without a valid timestamp received"),
("Invalid timestamp received on buffer"));
return GST_FLOW_ERROR;
invalid_segment:
{
const gchar *format = gst_format_get_name (padprivate->segment.format);
GST_OBJECT_UNLOCK (adder);
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (adder, STREAM, FAILED,
("This element only supports TIME segments, received other type"),
("Received a segment of type %s, only support time segment", format));
return GST_FLOW_ERROR;
}
}