/*
- Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
+ 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/>.
*/
#include "audio_decoder.h"
#include "resampler.h"
#include "util.h"
+#include "film.h"
+#include "log.h"
+#include "audio_content.h"
+#include "compose.hpp"
#include <iostream>
#include "i18n.h"
using boost::optional;
using boost::shared_ptr;
-AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, AudioDecoder* decoder)
+AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, shared_ptr<Log> log)
: _content (content)
, _stream (stream)
, _decoder (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_audio_frame_rate() != _stream->frame_rate()) {
- _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels ()));
+ if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
+ _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;
-
- 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_audio_frame_rate()), accurate);
+
+ /* If we are less than (about) 5 seconds behind the data that we want we'll
+ run through it rather than seeking.
+ */
+ Frame const slack = 5 * 48000;
+
+ if (frame < _decoded.frame || end > (_decoded.frame + _decoded.audio->frames() + slack)) {
+ /* Either we have no decoded data, all our data is after the time that we
+ want, or what we do have is a long way from what we want: seek */
+ _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
}
/* Offset of the data that we want from the start of _decoded.audio
(to be set up shortly)
*/
Frame decoded_offset = 0;
-
+
/* Now enough pass() calls will either:
* (a) give us what we want, or
* (b) hit the end of the decoder.
/* 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) &&
- !_decoder->pass ()
+ !_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),
+ LogEntry::TYPE_DEBUG_DECODE
+ );
} else {
while (
_decoded.audio->frames() < length &&
- !_decoder->pass ()
+ !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
)
{}
-
+
/* Use decoded_offset of 0, as we don't really care what frames we return */
}
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);
+
if (_resampler) {
data = _resampler->run (data);
}
- Frame const frame_rate = _content->resampled_audio_frame_rate ();
+ Frame const frame_rate = _content->resampled_frame_rate ();
if (_seek_reference) {
/* We've had an accurate seek and now we're seeing some data */
ContentTime const delta = time - _seek_reference.get ();
- Frame const delta_frames = delta.frames (frame_rate);
+ Frame const delta_frames = delta.frames_round (frame_rate);
if (delta_frames > 0) {
/* This data comes after the seek time. Pad the data with some silence. */
shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
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> ();
}
if (!_position) {
- _position = time.frames (frame_rate);
+ _position = time.frames_round (frame_rate);
}
DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
*/
return;
}
-
+
/* Resize _decoded to fit the new data */
int new_size = 0;
if (_decoded.audio->frames() == 0) {
/* Otherwise we need to extend _decoded to include the new stuff */
new_size = _position.get() + data->frames() - _decoded.frame;
}
-
+
_decoded.audio->ensure_size (new_size);
_decoded.audio->set_frames (new_size);
_position = _position.get() + data->frames ();
/* Limit the amount of data we keep in case nobody is asking for it */
- int const max_frames = _content->resampled_audio_frame_rate () * 10;
+ int const max_frames = _content->resampled_frame_rate () * 10;
if (_decoded.audio->frames() > max_frames) {
int const to_remove = _decoded.audio->frames() - max_frames;
_decoded.frame += to_remove;
_seek_reference = t;
}
}
+
+void
+AudioDecoderStream::set_fast ()
+{
+ if (_resampler) {
+ _resampler->set_fast ();
+ }
+}
projects / dcpomatic.git / blobdiff