/*
- 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 <iostream>
+#ifdef DCPOMATIC_WINDOWS
+#include <windows.h>
+#define ENABLE_SNDFILE_WINDOWS_PROTOTYPES 1
+#endif
#include <sndfile.h>
#include "sndfile_content.h"
#include "sndfile_decoder.h"
-#include "film.h"
#include "exceptions.h"
#include "audio_buffers.h"
using std::cout;
using boost::shared_ptr;
-SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
- : Decoder (f)
- , AudioDecoder (f, c)
- , _sndfile_content (c)
+SndfileDecoder::SndfileDecoder (shared_ptr<const SndfileContent> c)
+ : Sndfile (c)
+ , AudioDecoder (c)
+ , _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 ()
{
- sf_close (_sndfile);
delete[] _deinterleave_buffer;
}
-void
-SndfileDecoder::pass ()
+bool
+SndfileDecoder::pass (PassReason)
{
+ 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->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);
- int const channels = _sndfile_content->audio_channels ();
+ int const channels = _sndfile_content->audio_stream()->channels ();
- shared_ptr<AudioBuffers> audio (new AudioBuffers (channels, this_time));
+ shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
- if (_sndfile_content->audio_channels() == 1) {
+ if (_sndfile_content->audio_stream()->channels() == 1) {
/* No de-interleaving required */
- sf_read_float (_sndfile, audio->data(0), this_time);
+ sf_read_float (_sndfile, data->data(0), this_time);
} else {
/* Deinterleave */
if (!_deinterleave_buffer) {
sf_readf_float (_sndfile, _deinterleave_buffer, this_time);
vector<float*> out_ptr (channels);
for (int i = 0; i < channels; ++i) {
- out_ptr[i] = audio->data(i);
+ out_ptr[i] = data->data(i);
}
float* in_ptr = _deinterleave_buffer;
for (int i = 0; i < this_time; ++i) {
}
}
- audio->set_frames (this_time);
- Audio (audio, double(_done) / audio_frame_rate());
+ data->set_frames (this_time);
+ audio (_sndfile_content->audio_stream (), data, ContentTime::from_frames (_done, _info.samplerate));
_done += this_time;
_remaining -= this_time;
-}
-int
-SndfileDecoder::audio_channels () const
-{
- return _info.channels;
-}
-
-ContentAudioFrame
-SndfileDecoder::audio_length () const
-{
- return _info.frames;
+ return _remaining == 0;
}
-int
-SndfileDecoder::audio_frame_rate () const
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
{
- return _info.samplerate;
-}
+ AudioDecoder::seek (t, accurate);
-Time
-SndfileDecoder::next () const
-{
- return _next_audio;
+ _done = t.frames (_info.samplerate);
+ _remaining = _info.frames - _done;
}