/*
- 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 "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, 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_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 ();
_decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
}
-ContentAudio
-AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
-{
- shared_ptr<ContentAudio> dec;
-
- _content->film()->log()->log (String::compose ("ADS has request for %1 %2", frame, length), Log::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);
- }
-
- /* 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.
- *
- * If we are being accurate, we want the right frames,
- * otherwise any frames will do.
- */
- 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) &&
- !_decoder->pass ()
- )
- {}
-
- decoded_offset = frame - _decoded.frame;
- } else {
- while (
- _decoded.audio->frames() < length &&
- !_decoder->pass ()
- )
- {}
-
- /* Use decoded_offset of 0, as we don't really care what frames we return */
- }
-
- /* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
- if pass() returned true before we got enough data.
- */
- Frame const available = _decoded.audio->frames() - decoded_offset;
-
- /* We will return either that, or the requested amount, whichever is smaller */
- Frame const to_return = max ((Frame) 0, min (available, length));
-
- /* Copy our data to the output */
- shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
- out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
-
- Frame const remaining = max ((Frame) 0, available - to_return);
-
- /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
- _decoded.audio->move (decoded_offset + to_return, 0, remaining);
- /* And set up the number of frames we have left */
- _decoded.audio->set_frames (remaining);
- /* Also bump where those frames are in terms of the content */
- _decoded.frame += decoded_offset + to_return;
-
- return ContentAudio (out, frame);
-}
-
/** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
* We have to assume that we are feeding continuous data into the resampler, and so we get continuous
* data out. Hence we do the timestamping here, post-resampler, just by counting samples.
void
AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
- _content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), Log::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);
}
- 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 */
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> ();
}
_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;
}
void
-AudioDecoderStream::seek (ContentTime t, bool accurate)
+AudioDecoderStream::set_fast ()
{
- _position.reset ();
- reset_decoded ();
- if (accurate) {
- _seek_reference = t;
+ 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());
+}
projects / dcpomatic.git / blobdiff