/*
- Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
#include "audio_decoder.h"
#include "audio_buffers.h"
-#include "exceptions.h"
-#include "log.h"
+#include "audio_processor.h"
#include "resampler.h"
#include "util.h"
-#include "film.h"
-#include "audio_processor.h"
+#include <iostream>
#include "i18n.h"
-using std::stringstream;
using std::list;
using std::pair;
using std::cout;
}
shared_ptr<ContentAudio>
-AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+AudioDecoder::get_audio (Frame frame, Frame length, bool accurate)
{
shared_ptr<ContentAudio> dec;
- AudioFrame const end = frame + length - 1;
+ Frame const end = frame + length - 1;
if (frame < _decoded_audio.frame || end > (_decoded_audio.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, _audio_content->audio_frame_rate()), accurate);
+ seek (ContentTime::from_frames (frame, _audio_content->resampled_audio_frame_rate()), accurate);
}
/* Offset of the data that we want from the start of _decoded_audio.audio
(to be set up shortly)
*/
- AudioFrame decoded_offset = 0;
+ Frame decoded_offset = 0;
/* Now enough pass() calls will either:
* (a) give us what we want, or
*/
if (accurate) {
/* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */
- while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+ while ((_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end) && !pass (PASS_REASON_AUDIO)) {}
decoded_offset = frame - _decoded_audio.frame;
} else {
- while (!pass() && _decoded_audio.audio->frames() < length) {}
+ while (_decoded_audio.audio->frames() < length && !pass (PASS_REASON_AUDIO)) {}
/* Use decoded_offset of 0, as we don't really care what frames we return */
}
/* The amount of data available in _decoded_audio.audio starting from `frame'. This could be -ve
if pass() returned true before we got enough data.
*/
- AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
+ Frame const available = _decoded_audio.audio->frames() - decoded_offset;
/* We will return either that, or the requested amount, whichever is smaller */
- AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
+ Frame const to_return = max ((Frame) 0, min (available, length));
/* Copy our data to the output */
shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
- AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
+ 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.audio->move (decoded_offset + to_return, 0, remaining);
data = _processor->run (data);
}
- AudioFrame const frame_rate = _audio_content->resampled_audio_frame_rate ();
+ Frame const frame_rate = _audio_content->resampled_audio_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 ();
- AudioFrame const delta_frames = delta.frames (frame_rate);
+ Frame const delta_frames = delta.frames (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));
time -= delta;
} else if (delta_frames < 0) {
/* This data comes before the seek time. Throw some data away */
- AudioFrame const to_discard = min (-delta_frames, static_cast<AudioFrame> (data->frames()));
- AudioFrame const to_keep = data->frames() - to_discard;
+ 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.
_audio_position = time.frames (frame_rate);
}
- assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+ DCPOMATIC_ASSERT (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+ add (data);
+}
+void
+AudioDecoder::add (shared_ptr<const AudioBuffers> data)
+{
+ if (!_audio_position) {
+ /* This should only happen when there is a seek followed by a flush, but
+ we need to cope with it.
+ */
+ return;
+ }
+
/* Resize _decoded_audio to fit the new data */
int new_size = 0;
if (_decoded_audio.audio->frames() == 0) {
/* Copy new data in */
_decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
_audio_position = _audio_position.get() + data->frames ();
+
+ /* Limit the amount of data we keep in case nobody is asking for it */
+ int const max_frames = _audio_content->resampled_audio_frame_rate () * 10;
+ if (_decoded_audio.audio->frames() > max_frames) {
+ int const to_remove = _decoded_audio.audio->frames() - max_frames;
+ _decoded_audio.frame += to_remove;
+ _decoded_audio.audio->move (to_remove, 0, max_frames);
+ _decoded_audio.audio->set_frames (max_frames);
+ }
}
-/* XXX: called? */
void
AudioDecoder::flush ()
{
return;
}
- /*
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
- _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
- _audio_position = _audio_position.get() + b->frames ();
+ add (b);
}
- */
}
void
projects / dcpomatic.git / blobdiff