/*
- Copyright (C) 2012 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 <sndfile.h>
#include "sndfile_content.h"
#include "sndfile_decoder.h"
-#include "film.h"
#include "exceptions.h"
+#include "audio_buffers.h"
#include "i18n.h"
using std::cout;
using boost::shared_ptr;
-SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
- : Decoder (f)
- , AudioDecoder (f)
- , _sndfile_content (c)
+SndfileDecoder::SndfileDecoder (shared_ptr<const SndfileContent> c, bool fast)
+ : Sndfile (c)
+ , AudioDecoder (c, fast)
+ , _done (0)
+ , _remaining (_info.frames)
+ , _deinterleave_buffer (0)
{
- _sndfile = sf_open (_sndfile_content->file().string().c_str(), SFM_READ, &_info);
- if (!_sndfile) {
- throw DecodeError (_("could not open audio file for reading"));
- }
- _done = 0;
- _remaining = _info.frames;
}
SndfileDecoder::~SndfileDecoder ()
{
- if (_sndfile) {
- sf_close (_sndfile);
- }
+ delete[] _deinterleave_buffer;
}
bool
-SndfileDecoder::pass ()
+SndfileDecoder::pass (PassReason, bool)
{
+ if (_remaining == 0) {
+ return true;
+ }
+
/* Do things in half second blocks as I think there may be limits
to what FFmpeg (and in particular the resampler) can cope with.
*/
- sf_count_t const block = _sndfile_content->audio_frame_rate() / 2;
+ sf_count_t const block = _sndfile_content->audio_stream()->frame_rate() / 2;
sf_count_t const this_time = min (block, _remaining);
-
- shared_ptr<AudioBuffers> audio (new AudioBuffers (_sndfile_content->audio_channels(), this_time));
- sf_read_float (_sndfile, audio->data(0), this_time);
- audio->set_frames (this_time);
- Audio (audio, double(_done) / audio_frame_rate());
+
+ int const channels = _sndfile_content->audio_stream()->channels ();
+
+ shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
+
+ if (_sndfile_content->audio_stream()->channels() == 1) {
+ /* No de-interleaving required */
+ sf_read_float (_sndfile, data->data(0), this_time);
+ } else {
+ /* Deinterleave */
+ if (!_deinterleave_buffer) {
+ _deinterleave_buffer = new float[block * channels];
+ }
+ sf_readf_float (_sndfile, _deinterleave_buffer, this_time);
+ vector<float*> out_ptr (channels);
+ for (int i = 0; i < channels; ++i) {
+ out_ptr[i] = data->data(i);
+ }
+ float* in_ptr = _deinterleave_buffer;
+ for (int i = 0; i < this_time; ++i) {
+ for (int j = 0; j < channels; ++j) {
+ *out_ptr[j]++ = *in_ptr++;
+ }
+ }
+ }
+
+ data->set_frames (this_time);
+ audio (_sndfile_content->audio_stream (), data, ContentTime::from_frames (_done, _info.samplerate));
_done += this_time;
_remaining -= this_time;
- return (_remaining == 0);
+ return _remaining == 0;
}
-int
-SndfileDecoder::audio_channels () const
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
{
- return _info.channels;
-}
+ AudioDecoder::seek (t, accurate);
-ContentAudioFrame
-SndfileDecoder::audio_length () const
-{
- return _info.frames;
-}
-
-int
-SndfileDecoder::audio_frame_rate () const
-{
- return _info.samplerate;
+ _done = t.frames_round (_info.samplerate);
+ _remaining = _info.frames - _done;
}