/* Create sound output files with .tmp suffixes; we will rename
them if and when we complete.
*/
- for (int i = 0; i < _fs->audio_channels; ++i) {
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
SF_INFO sf_info;
- sf_info.samplerate = dcp_audio_sample_rate (_fs->audio_sample_rate);
+ sf_info.samplerate = dcp_audio_sample_rate (_fs->audio_sample_rate());
/* We write mono files */
sf_info.channels = 1;
sf_info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_24;
/* Only do the processing if we don't already have a file for this frame */
if (!boost::filesystem::exists (_opt->frame_out_path (frame, false))) {
- pair<string, string> const s = Filter::ffmpeg_strings (_fs->filters);
+ pair<string, string> const s = Filter::ffmpeg_strings (_fs->filters());
TIMING ("adding to queue of %1", _queue.size ());
_queue.push_back (boost::shared_ptr<DCPVideoFrame> (
new DCPVideoFrame (
- yuv, sub, _opt->out_size, _opt->padding, _fs->subtitle_offset, _fs->subtitle_scale,
- _fs->scaler, frame, _fs->frames_per_second, s.second,
+ yuv, sub, _opt->out_size, _opt->padding, _fs->subtitle_offset(), _fs->subtitle_scale(),
+ _fs->scaler(), frame, _fs->frames_per_second(), s.second,
Config::instance()->colour_lut_index (), Config::instance()->j2k_bandwidth (),
_log
)
void
J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format)
{
- if (_fs->audio_sample_rate != _fs->target_sample_rate ()) {
+ if (_fs->audio_sample_rate() != _fs->target_sample_rate()) {
#ifdef HAVE_SWRESAMPLE
stringstream s;
- s << "Will resample audio from " << _fs->audio_sample_rate << " to " << _fs->target_sample_rate();
+ s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_sample_rate();
_log->log (s.str ());
_swr_context = swr_alloc_set_opts (
_fs->target_sample_rate(),
audio_channel_layout,
audio_sample_format,
- _fs->audio_sample_rate,
+ _fs->audio_sample_rate(),
0, 0
);
if (_swr_context) {
while (1) {
- uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels];
+ uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels()];
uint8_t* out[2] = {
buffer,
0
break;
}
- write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels);
+ write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels());
}
swr_free (&_swr_context);
close_sound_files ();
/* Rename .wav.tmp files to .wav */
- for (int i = 0; i < _fs->audio_channels; ++i) {
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
if (boost::filesystem::exists (_opt->multichannel_audio_out_path (i, false))) {
boost::filesystem::remove (_opt->multichannel_audio_out_path (i, false));
}
/* And here's frames (where 1 frame is a collection of samples, 1 for each channel,
so for 5.1 a frame would be 6 samples)
*/
- int const frames = samples / _fs->audio_channels;
+ int const frames = samples / _fs->audio_channels();
/* Compute the resampled frame count and add 32 for luck */
- int const out_buffer_size_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate) + 32;
- int const out_buffer_size_bytes = out_buffer_size_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ int const out_buffer_size_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
+ int const out_buffer_size_bytes = out_buffer_size_frames * _fs->audio_channels() * _fs->bytes_per_sample();
out_buffer = new uint8_t[out_buffer_size_bytes];
uint8_t* out[2] = {
/* And point our variables at the resampled audio */
data = out_buffer;
- size = out_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ size = out_frames * _fs->audio_channels() * _fs->bytes_per_sample();
}
#endif
of the sample size and that size is a multiple of _fs->audio_channels * sample_size.
*/
- assert ((size % (_fs->audio_channels * _fs->bytes_per_sample())) == 0);
+ assert ((size % (_fs->audio_channels() * _fs->bytes_per_sample())) == 0);
assert ((_deinterleave_buffer_size % _fs->bytes_per_sample()) == 0);
/* XXX: this code is very tricksy and it must be possible to make it simpler ... */
int position = 0;
while (remaining > 0) {
/* How many bytes of the deinterleaved data to do this time */
- int this_time = min (remaining / _fs->audio_channels, _deinterleave_buffer_size);
- for (int i = 0; i < _fs->audio_channels; ++i) {
+ int this_time = min (remaining / _fs->audio_channels(), _deinterleave_buffer_size);
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
for (int j = 0; j < this_time; j += _fs->bytes_per_sample()) {
for (int k = 0; k < _fs->bytes_per_sample(); ++k) {
int const to = j + k;
- int const from = position + (i * _fs->bytes_per_sample()) + (j * _fs->audio_channels) + k;
+ int const from = position + (i * _fs->bytes_per_sample()) + (j * _fs->audio_channels()) + k;
_deinterleave_buffer[to] = data[from];
}
}
- switch (_fs->audio_sample_format) {
+ switch (_fs->audio_sample_format()) {
case AV_SAMPLE_FMT_S16:
sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / _fs->bytes_per_sample());
break;
}
position += this_time;
- remaining -= this_time * _fs->audio_channels;
+ remaining -= this_time * _fs->audio_channels();
}
}