if (c->subtitle) {
subtitle.reset (new SubtitleDecoder (this, c->subtitle, log));
}
+
+ _next_time.resize (_format_context->nb_streams);
}
void
of the block that do not form a complete sample or frame they will be dropped.
*/
int const total_samples = size / bytes_per_audio_sample (stream);
- int const frames = total_samples / stream->channels();
- shared_ptr<AudioBuffers> audio (new AudioBuffers (stream->channels(), frames));
+ int const channels = stream->channels();
+ int const frames = total_samples / channels;
+ shared_ptr<AudioBuffers> audio (new AudioBuffers (channels, frames));
+ float** data = audio->data();
switch (audio_sample_format (stream)) {
case AV_SAMPLE_FMT_U8:
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
- audio->data(channel)[sample] = float(*p++) / (1 << 23);
+ data[channel][sample] = float(*p++) / (1 << 23);
++channel;
- if (channel == stream->channels()) {
+ if (channel == channels) {
channel = 0;
++sample;
}
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
- audio->data(channel)[sample] = float(*p++) / (1 << 15);
+ data[channel][sample] = float(*p++) / (1 << 15);
++channel;
- if (channel == stream->channels()) {
+ if (channel == channels) {
channel = 0;
++sample;
}
case AV_SAMPLE_FMT_S16P:
{
int16_t** p = reinterpret_cast<int16_t **> (_frame->data);
- for (int i = 0; i < stream->channels(); ++i) {
+ for (int i = 0; i < channels; ++i) {
for (int j = 0; j < frames; ++j) {
- audio->data(i)[j] = static_cast<float>(p[i][j]) / (1 << 15);
+ data[i][j] = static_cast<float>(p[i][j]) / (1 << 15);
}
}
}
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
- audio->data(channel)[sample] = static_cast<float>(*p++) / 2147483648;
+ data[channel][sample] = static_cast<float>(*p++) / 2147483648;
++channel;
- if (channel == stream->channels()) {
+ if (channel == channels) {
channel = 0;
++sample;
}
case AV_SAMPLE_FMT_S32P:
{
int32_t** p = reinterpret_cast<int32_t **> (_frame->data);
- for (int i = 0; i < stream->channels(); ++i) {
+ for (int i = 0; i < channels; ++i) {
for (int j = 0; j < frames; ++j) {
- audio->data(i)[j] = static_cast<float>(p[i][j]) / 2147483648;
+ data[i][j] = static_cast<float>(p[i][j]) / 2147483648;
}
}
}
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
- audio->data(channel)[sample] = *p++;
+ data[channel][sample] = *p++;
++channel;
- if (channel == stream->channels()) {
+ if (channel == channels) {
channel = 0;
++sample;
}
float** p = reinterpret_cast<float**> (_frame->data);
/* Sometimes there aren't as many channels in the _frame as in the stream */
for (int i = 0; i < _frame->channels; ++i) {
- memcpy (audio->data(i), p[i], frames * sizeof(float));
+ memcpy (data[i], p[i], frames * sizeof(float));
}
- for (int i = _frame->channels; i < stream->channels(); ++i) {
+ for (int i = _frame->channels; i < channels; ++i) {
audio->make_silent (i);
}
}
*/
AVPacket copy_packet = _packet;
+ int const stream_index = copy_packet.stream_index;
/* XXX: inefficient */
vector<shared_ptr<FFmpegAudioStream> > streams = ffmpeg_content()->ffmpeg_audio_streams ();
vector<shared_ptr<FFmpegAudioStream> >::const_iterator stream = streams.begin ();
- while (stream != streams.end () && !(*stream)->uses_index (_format_context, copy_packet.stream_index)) {
+ while (stream != streams.end () && !(*stream)->uses_index (_format_context, stream_index)) {
++stream;
}
}
if (frame_finished) {
- ContentTime ct = ContentTime::from_seconds (
- av_frame_get_best_effort_timestamp (_frame) *
- av_q2d ((*stream)->stream (_format_context)->time_base))
- + _pts_offset;
-
shared_ptr<AudioBuffers> data = deinterleave_audio (*stream);
+ ContentTime ct;
+ if (_frame->pts == AV_NOPTS_VALUE && _next_time[stream_index]) {
+ /* In some streams we see not every frame coming through with a timestamp; for those
+ that have AV_NOPTS_VALUE we need to work out the timestamp ourselves. This is
+ particularly noticeable with TrueHD streams (see #1111).
+ */
+ ct = *_next_time[stream_index];
+ } else {
+ ct = ContentTime::from_seconds (
+ av_frame_get_best_effort_timestamp (_frame) *
+ av_q2d ((*stream)->stream (_format_context)->time_base))
+ + _pts_offset;
+ }
+
+ _next_time[stream_index] = ct + ContentTime::from_frames(data->frames(), (*stream)->frame_rate());
+
if (ct < ContentTime ()) {
/* Discard audio data that comes before time 0 */
Frame const remove = min (int64_t (data->frames()), (-ct).frames_ceil(double((*stream)->frame_rate ())));
}
if (ct < ContentTime()) {
- LOG_WARNING ("Crazy timestamp %1", to_string (ct));
+ LOG_WARNING (
+ "Crazy timestamp %1 for %2 samples in stream %3 packet pts %4 (ts=%5 tb=%6, off=%7)",
+ to_string(ct),
+ data->frames(),
+ copy_packet.stream_index,
+ copy_packet.pts,
+ av_frame_get_best_effort_timestamp(_frame),
+ av_q2d((*stream)->stream(_format_context)->time_base),
+ to_string(_pts_offset)
+ );
}
/* Give this data provided there is some, and its time is sane */
}
sub::RawSubtitle base;
- list<sub::RawSubtitle> raw = sub::SSAReader::parse_line (base, bits[9]);
+ list<sub::RawSubtitle> raw = sub::SSAReader::parse_line (
+ base,
+ bits[9],
+ _ffmpeg_content->video->size().width,
+ _ffmpeg_content->video->size().height
+ );
BOOST_FOREACH (sub::Subtitle const & i, sub::collect<list<sub::Subtitle> > (raw)) {
subtitle->emit_text_start (from, i);
projects / dcpomatic.git / blobdiff