/*
Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
- This program is free software; you can redistribute it and/or modify
+ This file is part of DCP-o-matic.
+
+ DCP-o-matic is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
- This program is distributed in the hope that it will be useful,
+ DCP-o-matic is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
- along with this program; if not, write to the Free Software
- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
using boost::optional;
using boost::shared_ptr;
-AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, shared_ptr<Log> log)
+AudioDecoderStream::AudioDecoderStream (
+ shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, AudioDecoder* audio_decoder, shared_ptr<Log> log
+ )
: _content (content)
, _stream (stream)
, _decoder (decoder)
+ , _audio_decoder (audio_decoder)
, _log (log)
+ /* We effectively start having done a seek to zero; this allows silence-padding of the first
+ data that comes out of our decoder.
+ */
+ , _seek_reference (ContentTime ())
{
if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
- _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), decoder->audio->fast ()));
+ _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ()));
}
reset_decoded ();
{
shared_ptr<ContentAudio> dec;
- _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
-
- Frame const end = frame + length - 1;
+ _log->log (
+ String::compose (
+ "ADS has request for %1 %2; has %3 %4",
+ frame, length, _decoded.frame, _decoded.audio->frames()
+ ), LogEntry::TYPE_DEBUG_DECODE
+ );
+
+ Frame const from = frame;
+ Frame const to = from + length;
+ Frame const have_from = _decoded.frame;
+ Frame const have_to = _decoded.frame + _decoded.audio->frames();
+
+ optional<Frame> missing;
+ if (have_from > from || have_to < to) {
+ /* We need something */
+ if (have_from <= from && from < have_to) {
+ missing = have_to;
+ } else {
+ missing = from;
+ }
+ }
- if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) {
- /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
- seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
+ if (missing) {
+ optional<ContentTime> pos = _audio_decoder->position ();
+ _log->log (
+ String::compose ("ADS suggests seek to %1 (now at %2)", *missing, pos ? to_string(pos.get()) : "none"),
+ LogEntry::TYPE_DEBUG_DECODE
+ );
+ _audio_decoder->maybe_seek (ContentTime::from_frames (*missing, _content->resampled_frame_rate()), accurate);
}
/* Offset of the data that we want from the start of _decoded.audio
if (accurate) {
/* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
while (
- (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
+ (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) <= to) &&
!_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
)
{}
decoded_offset = frame - _decoded.frame;
_log->log (
- String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
+ String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, have_from),
LogEntry::TYPE_DEBUG_DECODE
);
} else {
void
AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
- _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
+ _log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
if (_resampler) {
data = _resampler->run (data);
padded->copy_from (data.get(), data->frames(), 0, delta_frames);
data = padded;
time -= delta;
- } else if (delta_frames < 0) {
- /* This data comes before the seek time. Throw some data away */
- Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
- Frame const to_keep = data->frames() - to_discard;
- if (to_keep == 0) {
- /* We have to throw all this data away, so keep _seek_reference and
- try again next time some data arrives.
- */
- return;
- }
- shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
- trimmed->copy_from (data.get(), to_keep, to_discard, 0);
- data = trimmed;
- time += ContentTime::from_frames (to_discard, frame_rate);
}
_seek_reference = optional<ContentTime> ();
}
_seek_reference = t;
}
}
+
+void
+AudioDecoderStream::set_fast ()
+{
+ if (_resampler) {
+ _resampler->set_fast ();
+ }
+}
+
+optional<ContentTime>
+AudioDecoderStream::position () const
+{
+ if (!_position) {
+ return optional<ContentTime> ();
+ }
+
+ return ContentTime::from_frames (_position.get(), _content->resampled_frame_rate());
+}