本文整理汇总了C++中GST_BUFFER_OFFSET函数的典型用法代码示例。如果您正苦于以下问题:C++ GST_BUFFER_OFFSET函数的具体用法?C++ GST_BUFFER_OFFSET怎么用?C++ GST_BUFFER_OFFSET使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了GST_BUFFER_OFFSET函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: gstbt_audio_synth_create
static GstFlowReturn
gstbt_audio_synth_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstBtAudioSynth *src = GSTBT_AUDIO_SYNTH (basesrc);
GstBtAudioSynthClass *klass = GSTBT_AUDIO_SYNTH_GET_CLASS (src);
GstFlowReturn res;
GstBuffer *buf;
GstMapInfo info;
GstClockTime next_running_time, ticktime;
gint64 n_samples;
gdouble samples_done;
guint samples_per_buffer;
gboolean partial_buffer = FALSE;
if (G_UNLIKELY (src->eos_reached)) {
GST_WARNING_OBJECT (src, "EOS reached");
return GST_FLOW_EOS;
}
// the amount of samples to produce (handle rounding errors by collecting left over fractions)
samples_done =
(gdouble) src->running_time * (gdouble) src->info.rate /
(gdouble) GST_SECOND;
if (!src->reverse) {
samples_per_buffer =
(guint) (src->samples_per_buffer + (samples_done -
(gdouble) src->n_samples));
} else {
samples_per_buffer =
(guint) (src->samples_per_buffer + ((gdouble) src->n_samples -
samples_done));
}
GST_DEBUG_OBJECT (src,
"samples_done=%lf, src->n_samples=%" G_GUINT64_FORMAT
", src->n_samples_stop=%" G_GUINT64_FORMAT,
samples_done, src->n_samples, src->n_samples_stop);
GST_DEBUG_OBJECT (src, "samples-per-buffer=%7u (%8.3lf), length=%u",
samples_per_buffer, src->samples_per_buffer, length);
/* check for eos */
if (src->check_eos) {
if (!src->reverse) {
partial_buffer = ((src->n_samples_stop >= src->n_samples) &&
(src->n_samples_stop < src->n_samples + samples_per_buffer));
} else {
partial_buffer = ((src->n_samples_stop < src->n_samples) &&
(src->n_samples_stop >= src->n_samples - samples_per_buffer));
}
}
if (G_UNLIKELY (partial_buffer)) {
/* calculate only partial buffer */
if (!src->reverse) {
src->generate_samples_per_buffer =
(guint) (src->n_samples_stop - src->n_samples);
ticktime = src->stop_time - src->running_time;
} else {
src->generate_samples_per_buffer =
(guint) (src->n_samples - src->n_samples_stop);
ticktime = src->running_time - src->stop_time;
}
if (G_UNLIKELY (!src->generate_samples_per_buffer)) {
GST_WARNING_OBJECT (src, "0 samples left -> EOS reached");
src->eos_reached = TRUE;
return GST_FLOW_EOS;
}
n_samples = src->n_samples_stop;
src->eos_reached = TRUE;
GST_INFO_OBJECT (src, "partial buffer: %u, ticktime: %" GST_TIME_FORMAT,
src->generate_samples_per_buffer, GST_TIME_ARGS (ticktime));
} else {
/* calculate full buffer */
src->generate_samples_per_buffer = samples_per_buffer;
n_samples =
src->n_samples +
(src->reverse ? (-samples_per_buffer) : samples_per_buffer);
ticktime = src->ticktime;
}
next_running_time =
src->running_time + (src->reverse ? (-ticktime) : ticktime);
src->ticktime_err_accum =
src->ticktime_err_accum +
(src->reverse ? (-src->ticktime_err) : src->ticktime_err);
res = GST_BASE_SRC_GET_CLASS (basesrc)->alloc (basesrc, src->n_samples,
gstbt_audio_synth_calculate_buffer_size (src), &buf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
return res;
}
if (!src->reverse) {
GST_BUFFER_TIMESTAMP (buf) =
src->running_time + (GstClockTime) src->ticktime_err_accum;
GST_BUFFER_DURATION (buf) = next_running_time - src->running_time;
GST_BUFFER_OFFSET (buf) = src->n_samples;
GST_BUFFER_OFFSET_END (buf) = n_samples;
} else {
GST_BUFFER_TIMESTAMP (buf) =
next_running_time + (GstClockTime) src->ticktime_err_accum;
//.........这里部分代码省略.........
开发者ID:Buzztrax,项目名称:buzztrax,代码行数:101,代码来源:audiosynth.c
示例2: data_proc
static OSErr
data_proc (SGChannel c, Ptr p, long len, long *offset, long chRefCon,
TimeValue time, short writeType, long refCon)
{
GstOSXVideoSrc *self;
gint fps_n, fps_d;
GstClockTime duration, timestamp, latency;
CodecFlags flags;
ComponentResult err;
PixMapHandle hPixMap;
Rect portRect;
int pix_rowBytes;
void *pix_ptr;
int pix_height;
int pix_size;
self = GST_OSX_VIDEO_SRC (refCon);
if (self->buffer != NULL) {
gst_buffer_unref (self->buffer);
self->buffer = NULL;
}
err = DecompressSequenceFrameS (self->dec_seq, p, len, 0, &flags, NULL);
if (err != noErr) {
GST_ERROR_OBJECT (self, "DecompressSequenceFrameS returned %d", (int) err);
return err;
}
hPixMap = GetGWorldPixMap (self->world);
LockPixels (hPixMap);
GetPortBounds (self->world, &portRect);
pix_rowBytes = (int) GetPixRowBytes (hPixMap);
pix_ptr = GetPixBaseAddr (hPixMap);
pix_height = (portRect.bottom - portRect.top);
pix_size = pix_rowBytes * pix_height;
GST_DEBUG_OBJECT (self, "num=%5d, height=%d, rowBytes=%d, size=%d",
self->seq_num, pix_height, pix_rowBytes, pix_size);
fps_n = FRAMERATE;
fps_d = 1;
duration = gst_util_uint64_scale_int (GST_SECOND, fps_d, fps_n);
latency = duration;
timestamp = gst_clock_get_time (GST_ELEMENT_CAST (self)->clock);
timestamp -= gst_element_get_base_time (GST_ELEMENT_CAST (self));
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
self->buffer = gst_buffer_new_and_alloc (pix_size);
GST_BUFFER_OFFSET (self->buffer) = self->seq_num;
GST_BUFFER_TIMESTAMP (self->buffer) = timestamp;
memcpy (GST_BUFFER_DATA (self->buffer), pix_ptr, pix_size);
self->seq_num++;
UnlockPixels (hPixMap);
return noErr;
}
开发者ID:ChinnaSuhas,项目名称:ossbuild,代码行数:64,代码来源:osxvideosrc.c
示例3: gst_pad_probe_info_get_buffer
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*,
GstPadProbeInfo* info,
gpointer self) {
GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
GstBuffer* buf = gst_pad_probe_info_get_buffer(info);
QList<BufferConsumer*> consumers;
{
QMutexLocker l(&instance->buffer_consumers_mutex_);
consumers = instance->buffer_consumers_;
}
for (BufferConsumer* consumer : consumers) {
gst_buffer_ref(buf);
consumer->ConsumeBuffer(buf, instance->id());
}
// Calculate the end time of this buffer so we can stop playback if it's
// after the end time of this song.
if (instance->end_offset_nanosec_ > 0) {
quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
quint64 duration = GST_BUFFER_DURATION(buf);
quint64 end_time = start_time + duration;
if (end_time > instance->end_offset_nanosec_) {
if (instance->has_next_valid_url()) {
if (instance->next_url_ == instance->url_ &&
instance->next_beginning_offset_nanosec_ ==
instance->end_offset_nanosec_) {
// The "next" song is actually the next segment of this file - so
// cheat and keep on playing, but just tell the Engine we've moved on.
instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
instance->next_url_ = QUrl();
instance->next_beginning_offset_nanosec_ = 0;
instance->next_end_offset_nanosec_ = 0;
// GstEngine will try to seek to the start of the new section, but
// we're already there so ignore it.
instance->ignore_next_seek_ = true;
emit instance->EndOfStreamReached(instance->id(), true);
} else {
// We have a next song but we can't cheat, so move to it normally.
instance->TransitionToNext();
}
} else {
// There's no next song
emit instance->EndOfStreamReached(instance->id(), false);
}
}
}
if (instance->emit_track_ended_on_time_discontinuity_) {
if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLAG_DISCONT) ||
GST_BUFFER_OFFSET(buf) < instance->last_buffer_offset_) {
qLog(Debug) << "Buffer discontinuity - emitting EOS";
instance->emit_track_ended_on_time_discontinuity_ = false;
emit instance->EndOfStreamReached(instance->id(), true);
}
}
instance->last_buffer_offset_ = GST_BUFFER_OFFSET(buf);
return GST_PAD_PROBE_OK;
}
开发者ID:ivovegter,项目名称:Clementine,代码行数:64,代码来源:gstenginepipeline.cpp
示例4: gst_wavpack_dec_handle_frame
//.........这里部分代码省略.........
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
#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;
gst_wavpack_dec_negotiate (dec);
/* 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 */
dec_data = g_malloc (4 * wph.block_samples * dec->channels);
/* decode */
decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
outbuf = gst_buffer_new_and_alloc (unpacked_size);
/* legacy; pass along offset, whatever that might entail */
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET (buf);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = omap.data;
width = dec->width;
depth = dec->depth;
max = dec->channels * wph.block_samples;
if (width == 8) {
gint8 *outbuffer = (gint8 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint8) (dec_data[i + reorder_map[j]]);
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ = (gint16) (dec_data[i + reorder_map[j]]);
}
} else if (dec->width == 32) {
gint32 *outbuffer = (gint32 *) out_data;
gint *reorder_map = dec->channel_reorder_map;
if (width != depth) {
for (i = 0; i < max; i += dec->channels) {
for (j = 0; j < dec->channels; j++)
*outbuffer++ =
(gint32) (dec_data[i + reorder_map[j]] << (width - depth));
开发者ID:Lachann,项目名称:gst-plugins-good,代码行数:67,代码来源:gstwavpackdec.c
示例5: gst_jif_mux_recombine_image
static GstFlowReturn
gst_jif_mux_recombine_image (GstJifMux * self, GstBuffer ** new_buf,
GstBuffer * old_buf)
{
GstBuffer *buf;
GstByteWriter *writer;
GstFlowReturn fret;
GstJifMuxMarker *m;
GList *node;
guint size = self->priv->scan_size;
gboolean writer_status = TRUE;
/* iterate list and collect size */
for (node = self->priv->markers; node; node = g_list_next (node)) {
m = (GstJifMuxMarker *) node->data;
/* some markers like e.g. SOI are empty */
if (m->size) {
size += 2 + m->size;
}
/* 0xff <marker> */
size += 2;
}
GST_INFO_OBJECT (self, "old size: %u, new size: %u",
GST_BUFFER_SIZE (old_buf), size);
/* allocate new buffer */
fret = gst_pad_alloc_buffer_and_set_caps (self->priv->srcpad,
GST_BUFFER_OFFSET (old_buf), size, GST_PAD_CAPS (self->priv->srcpad),
&buf);
if (fret != GST_FLOW_OK)
goto no_buffer;
/* copy buffer metadata */
gst_buffer_copy_metadata (buf, old_buf,
GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS);
/* memcopy markers */
writer = gst_byte_writer_new_with_buffer (buf, TRUE);
for (node = self->priv->markers; node && writer_status;
node = g_list_next (node)) {
m = (GstJifMuxMarker *) node->data;
writer_status &= gst_byte_writer_put_uint8 (writer, 0xff);
writer_status &= gst_byte_writer_put_uint8 (writer, m->marker);
GST_DEBUG_OBJECT (self, "marker = %2x, size = %u", m->marker, m->size + 2);
if (m->size) {
writer_status &= gst_byte_writer_put_uint16_be (writer, m->size + 2);
writer_status &= gst_byte_writer_put_data (writer, m->data, m->size);
}
if (m->marker == SOS) {
GST_DEBUG_OBJECT (self, "scan data, size = %u", self->priv->scan_size);
writer_status &=
gst_byte_writer_put_data (writer, self->priv->scan_data,
self->priv->scan_size);
}
}
gst_byte_writer_free (writer);
if (!writer_status) {
GST_WARNING_OBJECT (self, "Failed to write to buffer, calculated size "
"was probably too short");
g_assert_not_reached ();
}
*new_buf = buf;
return GST_FLOW_OK;
no_buffer:
GST_WARNING_OBJECT (self, "failed to allocate output buffer, flow_ret = %s",
gst_flow_get_name (fret));
return fret;
}
开发者ID:neduartes,项目名称:gst-plugins-bad,代码行数:76,代码来源:gstjifmux.c
示例6: do_perfect_stream_test
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_static_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));
//.........这里部分代码省略.........
开发者ID:ChinnaSuhas,项目名称:ossbuild,代码行数:101,代码来源:audiorate.c
示例7: gst_buffer_create_sub
/**
* gst_buffer_create_sub:
* @parent: a #GstBuffer.
* @offset: the offset into parent #GstBuffer at which the new sub-buffer
* begins.
* @size: the size of the new #GstBuffer sub-buffer, in bytes.
*
* Creates a sub-buffer from @parent at @offset and @size.
* This sub-buffer uses the actual memory space of the parent buffer.
* This function will copy the offset and timestamp fields when the
* offset is 0. If not, they will be set to #GST_CLOCK_TIME_NONE and
* #GST_BUFFER_OFFSET_NONE.
* If @offset equals 0 and @size equals the total size of @buffer, the
* duration and offset end fields are also copied. If not they will be set
* to #GST_CLOCK_TIME_NONE and #GST_BUFFER_OFFSET_NONE.
*
* MT safe.
* Returns: the new #GstBuffer.
* Returns NULL if the arguments were invalid.
*/
GstBuffer *
gst_buffer_create_sub (GstBuffer * buffer, guint offset, guint size)
{
GstBuffer *subbuffer;
GstBuffer *parent;
gboolean complete;
g_return_val_if_fail (buffer != NULL, NULL);
g_return_val_if_fail (buffer->mini_object.refcount > 0, NULL);
g_return_val_if_fail (buffer->size >= offset + size, NULL);
/* find real parent */
if (GST_IS_SUBBUFFER (buffer)) {
parent = buffer->parent;
} else {
parent = buffer;
}
gst_buffer_ref (parent);
/* create the new buffer */
subbuffer = gst_buffer_new ();
subbuffer->parent = parent;
GST_BUFFER_FLAG_SET (subbuffer, GST_BUFFER_FLAG_READONLY);
GST_CAT_LOG (GST_CAT_BUFFER, "new subbuffer %p (parent %p)", subbuffer,
parent);
/* set the right values in the child */
GST_BUFFER_DATA (subbuffer) = buffer->data + offset;
GST_BUFFER_SIZE (subbuffer) = size;
if ((offset == 0) && (size == GST_BUFFER_SIZE (buffer))) {
/* copy all the flags except IN_CAPS */
GST_BUFFER_FLAG_SET (subbuffer, GST_BUFFER_FLAGS (buffer));
GST_BUFFER_FLAG_UNSET (subbuffer, GST_BUFFER_FLAG_IN_CAPS);
} else {
/* copy only PREROLL & GAP flags */
GST_BUFFER_FLAG_SET (subbuffer, (GST_BUFFER_FLAGS (buffer) &
(GST_BUFFER_FLAG_PREROLL | GST_BUFFER_FLAG_GAP)));
}
/* we can copy the timestamp and offset if the new buffer starts at
* offset 0 */
if (offset == 0) {
GST_BUFFER_TIMESTAMP (subbuffer) = GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_OFFSET (subbuffer) = GST_BUFFER_OFFSET (buffer);
complete = (buffer->size == size);
} else {
GST_BUFFER_TIMESTAMP (subbuffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET (subbuffer) = GST_BUFFER_OFFSET_NONE;
complete = FALSE;
}
if (complete) {
GstCaps *caps;
/* if we copied the complete buffer we can copy the duration,
* offset_end and caps as well */
GST_BUFFER_DURATION (subbuffer) = GST_BUFFER_DURATION (buffer);
GST_BUFFER_OFFSET_END (subbuffer) = GST_BUFFER_OFFSET_END (buffer);
if ((caps = GST_BUFFER_CAPS (buffer)))
gst_caps_ref (caps);
GST_BUFFER_CAPS (subbuffer) = caps;
} else {
GST_BUFFER_DURATION (subbuffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (subbuffer) = GST_BUFFER_OFFSET_NONE;
GST_BUFFER_CAPS (subbuffer) = NULL;
}
return subbuffer;
}
开发者ID:zsx,项目名称:ossbuild,代码行数:90,代码来源:gstbuffer.c
示例8: gst_interleave_collected
static GstFlowReturn
gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
{
guint size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
GSList *collected;
guint nsamples;
guint ncollected = 0;
gboolean empty = TRUE;
gint width = self->width / 8;
g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED);
size = gst_collect_pads_available (pads);
g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR);
GST_DEBUG_OBJECT (self, "Starting to collect %u bytes from %d channels", size,
self->channels);
nsamples = size / width;
ret =
gst_pad_alloc_buffer (self->src, GST_BUFFER_OFFSET_NONE,
size * self->channels, GST_PAD_CAPS (self->src), &outbuf);
if (ret != GST_FLOW_OK) {
return ret;
} else if (outbuf == NULL || GST_BUFFER_SIZE (outbuf) < size * self->channels) {
gst_buffer_unref (outbuf);
return GST_FLOW_NOT_NEGOTIATED;
} else if (!gst_caps_is_equal (GST_BUFFER_CAPS (outbuf),
GST_PAD_CAPS (self->src))) {
gst_buffer_unref (outbuf);
return GST_FLOW_NOT_NEGOTIATED;
}
memset (GST_BUFFER_DATA (outbuf), 0, size * self->channels);
for (collected = pads->data; collected != NULL; collected = collected->next) {
GstCollectData *cdata;
GstBuffer *inbuf;
guint8 *outdata;
cdata = (GstCollectData *) collected->data;
inbuf = gst_collect_pads_take_buffer (pads, cdata, size);
if (inbuf == NULL) {
GST_DEBUG_OBJECT (cdata->pad, "No buffer available");
goto next;
}
ncollected++;
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))
goto next;
empty = FALSE;
outdata =
GST_BUFFER_DATA (outbuf) +
width * GST_INTERLEAVE_PAD_CAST (cdata->pad)->channel;
self->func (outdata, GST_BUFFER_DATA (inbuf), self->channels, nsamples);
next:
if (inbuf)
gst_buffer_unref (inbuf);
}
if (ncollected == 0)
goto eos;
if (self->segment_pending) {
GstEvent *event;
event = gst_event_new_new_segment_full (FALSE, self->segment_rate,
1.0, GST_FORMAT_TIME, self->timestamp, -1, self->segment_position);
gst_pad_push_event (self->src, event);
self->segment_pending = FALSE;
self->segment_position = 0;
}
GST_BUFFER_TIMESTAMP (outbuf) = self->timestamp;
GST_BUFFER_OFFSET (outbuf) = self->offset;
self->offset += nsamples;
self->timestamp = gst_util_uint64_scale_int (self->offset,
//.........这里部分代码省略.........
开发者ID:prajnashi,项目名称:gst-plugins-good,代码行数:101,代码来源:interleave.c
示例9: gst_wavpack_parse_push_buffer
static GstFlowReturn
gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf,
WavpackHeader * header)
{
wvparse->current_offset += header->ckSize + 8;
wvparse->segment.last_stop = header->block_index;
if (wvparse->need_newsegment) {
if (gst_wavpack_parse_send_newsegment (wvparse, FALSE))
wvparse->need_newsegment = FALSE;
}
/* send any queued events */
if (wvparse->queued_events) {
GList *l;
for (l = wvparse->queued_events; l != NULL; l = l->next) {
gst_pad_push_event (wvparse->srcpad, GST_EVENT (l->data));
}
g_list_free (wvparse->queued_events);
wvparse->queued_events = NULL;
}
if (wvparse->pending_buffer == NULL) {
wvparse->pending_buffer = buf;
wvparse->pending_offset = header->block_index;
} else if (wvparse->pending_offset == header->block_index) {
wvparse->pending_buffer = gst_buffer_join (wvparse->pending_buffer, buf);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (wvparse->pending_buffer);
wvparse->pending_buffer = buf;
wvparse->pending_offset = header->block_index;
}
if (!(header->flags & FINAL_BLOCK))
return GST_FLOW_OK;
buf = wvparse->pending_buffer;
wvparse->pending_buffer = NULL;
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header->block_index,
GST_SECOND, wvparse->samplerate);
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header->block_samples,
GST_SECOND, wvparse->samplerate);
GST_BUFFER_OFFSET (buf) = header->block_index;
GST_BUFFER_OFFSET_END (buf) = header->block_index + header->block_samples;
if (wvparse->discont || wvparse->next_block_index != header->block_index) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
wvparse->discont = FALSE;
}
wvparse->next_block_index = header->block_index + header->block_samples;
gst_buffer_set_caps (buf, GST_PAD_CAPS (wvparse->srcpad));
GST_LOG_OBJECT (wvparse, "Pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
return gst_pad_push (wvparse->srcpad, buf);
}
开发者ID:JJCG,项目名称:gst-plugins-good,代码行数:63,代码来源:gstwavpackparse.c
示例10: gst_decklink_src_task
static void
gst_decklink_src_task (void *priv)
{
GstDecklinkSrc *decklinksrc = GST_DECKLINK_SRC (priv);
GstBuffer *buffer;
GstBuffer *audio_buffer;
IDeckLinkVideoInputFrame *video_frame;
IDeckLinkAudioInputPacket *audio_frame;
void *data;
int n_samples;
GstFlowReturn ret;
const GstDecklinkMode *mode;
GST_DEBUG_OBJECT (decklinksrc, "task");
g_mutex_lock (decklinksrc->mutex);
while (decklinksrc->video_frame == NULL && !decklinksrc->stop) {
g_cond_wait (decklinksrc->cond, decklinksrc->mutex);
}
video_frame = decklinksrc->video_frame;
audio_frame = decklinksrc->audio_frame;
decklinksrc->video_frame = NULL;
decklinksrc->audio_frame = NULL;
g_mutex_unlock (decklinksrc->mutex);
if (decklinksrc->stop) {
GST_DEBUG ("stopping task");
return;
}
/* warning on dropped frames */
if (decklinksrc->dropped_frames - decklinksrc->dropped_frames_old > 0) {
GST_ELEMENT_WARNING (decklinksrc, RESOURCE, READ,
("Dropped %d frame(s), for a total of %d frame(s)",
decklinksrc->dropped_frames - decklinksrc->dropped_frames_old,
decklinksrc->dropped_frames), (NULL));
decklinksrc->dropped_frames_old = decklinksrc->dropped_frames;
}
mode = gst_decklink_get_mode (decklinksrc->mode);
video_frame->GetBytes (&data);
if (decklinksrc->copy_data) {
buffer = gst_buffer_new_and_alloc (mode->width * mode->height * 2);
memcpy (GST_BUFFER_DATA (buffer), data, mode->width * mode->height * 2);
video_frame->Release ();
} else {
buffer = gst_buffer_new ();
GST_BUFFER_SIZE (buffer) = mode->width * mode->height * 2;
GST_BUFFER_DATA (buffer) = (guint8 *) data;
GST_BUFFER_FREE_FUNC (buffer) = video_frame_free;
GST_BUFFER_MALLOCDATA (buffer) = (guint8 *) video_frame;
}
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (decklinksrc->frame_num * GST_SECOND,
mode->fps_d, mode->fps_n);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int ((decklinksrc->frame_num + 1) * GST_SECOND,
mode->fps_d, mode->fps_n) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_OFFSET (buffer) = decklinksrc->frame_num;
GST_BUFFER_OFFSET_END (buffer) = decklinksrc->frame_num;
if (decklinksrc->frame_num == 0) {
GstEvent *event;
gboolean ret;
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0,
GST_CLOCK_TIME_NONE, 0);
if (gst_pad_is_linked (decklinksrc->videosrcpad)) {
gst_event_ref (event);
ret = gst_pad_push_event (decklinksrc->videosrcpad, event);
if (!ret) {
GST_ERROR_OBJECT (decklinksrc, "new segment event ret=%d", ret);
gst_event_unref (event);
return;
}
} else {
gst_event_unref (event);
}
if (gst_pad_is_linked (decklinksrc->audiosrcpad)) {
ret = gst_pad_push_event (decklinksrc->audiosrcpad, event);
if (!ret) {
GST_ERROR_OBJECT (decklinksrc, "new segment event ret=%d", ret);
gst_event_unref (event);
}
} else {
gst_event_unref (event);
}
}
if (decklinksrc->video_caps == NULL) {
//.........这里部分代码省略.........
开发者ID:drothlis,项目名称:gst-plugins-bad,代码行数:101,代码来源:gstdecklinksrc.cpp
示例11: gst_identity_transform_ip
static GstFlowReturn
gst_identity_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstIdentity *identity = GST_IDENTITY (trans);
GstClockTime rundts = GST_CLOCK_TIME_NONE;
GstClockTime runpts = GST_CLOCK_TIME_NONE;
GstClockTime ts, duration, runtimestamp;
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_PTS (buf) = GST_BUFFER_DTS (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) {
rundts = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_DTS (buf));
runpts = gst_segment_to_running_time (&trans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buf));
}
if (GST_CLOCK_TIME_IS_VALID (rundts))
runtimestamp = rundts;
else if (GST_CLOCK_TIME_IS_VALID (runpts))
runtimestamp = runpts;
else
runtimestamp = 0;
ret = gst_identity_do_sync (identity, runtimestamp);
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_DTS (buf) = rundts;
GST_BUFFER_PTS (buf) = runpts;
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);
}
//.........这里部分代码省略.........
开发者ID:mparis,项目名称:gstreamer,代码行数:101,代码来源:gstidentity.c
示例12: gst_vdp_mpeg_dec_decode
static GstFlowReturn
gst_vdp_mpeg_dec_decode (GstVdpMpegDec * mpeg_dec,
GstClockTime timestamp, gint64 size)
{
VdpPictureInfoMPEG1Or2 *info;
GstBuffer *buffer;
GstBuffer *outbuf;
VdpVideoSurface surface;
GstVdpDevice *device;
VdpBitstreamBuffer vbit[1];
VdpStatus status;
info = &mpeg_dec->vdp_info;
if (info->picture_coding_type != B_FRAME) {
if (info->backward_reference != VDP_INVALID_HANDLE) {
gst_buffer_ref (mpeg_dec->b_buffer);
gst_vdp_mpeg_dec_push_video_buffer (mpeg_dec,
GST_VDP_VIDEO_BUFFER (mpeg_dec->b_buffer));
}
if (info->forward_reference != VDP_INVALID_HANDLE) {
gst_buffer_unref (mpeg_dec->f_buffer);
info->forward_reference = VDP_INVALID_HANDLE;
}
info->forward_reference = info->backward_reference;
mpeg_dec->f_buffer = mpeg_dec->b_buffer;
info->backward_reference = VDP_INVALID_HANDLE;
}
if (gst_vdp_mpeg_dec_alloc_buffer (mpeg_dec, &outbuf) != GST_FLOW_OK) {
gst_adapter_clear (mpeg_dec->adapter);
return GST_FLOW_ERROR;
}
device = GST_VDP_VIDEO_BUFFER (outbuf)->device;
if (info->forward_reference != VDP_INVALID_HANDLE &&
info->picture_coding_type != I_FRAME)
gst_vdp_video_buffer_add_reference (GST_VDP_VIDEO_BUFFER (outbuf),
GST_VDP_VIDEO_BUFFER (mpeg_dec->f_buffer));
if (info->backward_reference != VDP_INVALID_HANDLE
&& info->picture_coding_type == B_FRAME)
gst_vdp_video_buffer_add_reference (GST_VDP_VIDEO_BUFFER (outbuf),
GST_VDP_VIDEO_BUFFER (mpeg_dec->b_buffer));
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = mpeg_dec->duration;
GST_BUFFER_OFFSET (outbuf) = mpeg_dec->frame_nr;
GST_BUFFER_SIZE (outbuf) = size;
if (info->picture_coding_type == I_FRAME)
GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
else
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
if (info->top_field_first)
GST_BUFFER_FLAG_SET (outbuf, GST_VIDEO_BUFFER_TFF);
else
GST_BUFFER_FLAG_UNSET (outbuf, GST_VIDEO_BUFFER_TFF);
buffer = gst_adapter_take_buffer (mpeg_dec->adapter,
gst_adapter_available (mpeg_dec->adapter));
surface = GST_VDP_VIDEO_BUFFER (outbuf)->surface;
vbit[0].struct_version = VDP_BITSTREAM_BUFFER_VERSION;
vbit[0].bitstream = GST_BUFFER_DATA (buffer);
vbit[0].bitstream_bytes = GST_BUFFER_SIZE (buffer);
status = device->vdp_decoder_render (mpeg_dec->decoder, surface,
(VdpPictureInfo *) info, 1, vbit);
gst_buffer_unref (buffer);
info->slice_count = 0;
if (status != VDP_STATUS_OK) {
GST_ELEMENT_ERROR (mpeg_dec, RESOURCE, READ,
("Could not decode"),
("Error returned from vdpau was: %s",
device->vdp_get_error_string (status)));
gst_buffer_unref (GST_BUFFER (outbuf));
return GST_FLOW_ERROR;
}
if (info->picture_coding_type == B_FRAME) {
gst_vdp_mpeg_dec_push_video_buffer (mpeg_dec,
GST_VDP_VIDEO_BUFFER (outbuf));
} else {
info->backward_reference = surface;
mpeg_dec->b_buffer = GST_BUFFER (outbuf);
}
return GST_FLOW_OK;
}
开发者ID:zsx,项目名称:ossbuild,代码行数:99,代码来源:gstvdpmpegdec.c
示例13: gst_identity_transform_ip
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_PTS (buf) = GST_BUFFER_DTS (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_PTS (buf) = GST_BUFFER_DTS (buf) = runtimestamp;
GST_BUFFER_OFFSET (buf) = GST_CLOCK_TIME_NONE;
GST_BUFFER_OFFSET_END (buf) = GST_CLOCK_TIME_NONE;
}
return ret;
//.........这里部分代码省略.........
开发者ID:sebras,项目名称:gstreamer,代码行数:101,代码来源:gstidentity.c
示例14: gst_inter_video_src_create
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;
}
开发者ID:jcaden,项目名称:gst-plugins-bad,代码行数:64,代码来源:gstintervideosrc.c
示例15: gst_fake_sink_render
static GstFlowReturn
gst_fake_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstFakeSink *sink = GST_FAKE_SINK_CAST (bsink);
if (sink->num_buffers_left == 0)
goto eos;
if (sink->num_buffers_left != -1)
sink->num_buffers_left--;
if (!sink->silent) {
gchar dts_str[64], pts_str[64], dur_str[64];
gchar flag_str[100];
GST_OBJECT_LOCK (sink);
g_free (sink->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));
}
{
const char *flag_list[15] = {
"",
|
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