#ifdef HAVE_SWRESAMPLE
, _swr_context (0)
#endif
- , _deinterleave_buffer_size (8192)
- , _deinterleave_buffer (0)
, _process_end (false)
{
/* Create sound output files with .tmp suffixes; we will rename
}
_sound_files.push_back (f);
}
-
- /* Create buffer for deinterleaving audio */
- _deinterleave_buffer = new uint8_t[_deinterleave_buffer_size];
}
J2KWAVEncoder::~J2KWAVEncoder ()
{
terminate_worker_threads ();
- delete[] _deinterleave_buffer;
close_sound_files ();
}
stringstream s;
s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_sample_rate();
_log->log (s.str ());
-
+
+ /* We will be using planar float data when we call the resampler */
_swr_context = swr_alloc_set_opts (
0,
audio_channel_layout,
- audio_sample_format,
+ AV_SAMPLE_FMT_FLTP,
_fs->target_sample_rate(),
audio_channel_layout,
- audio_sample_format,
+ AV_SAMPLE_FMT_FLTP,
_fs->audio_sample_rate(),
0, 0
);
#if HAVE_SWRESAMPLE
if (_swr_context) {
+ float* out[_fs->audio_channels()];
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ out[i] = new float[256];
+ }
+
while (1) {
- uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels()];
- uint8_t* out[2] = {
- buffer,
- 0
- };
-
- int const frames = swr_convert (_swr_context, out, 256, 0, 0);
+ int const frames = swr_convert (_swr_context, (uint8_t **) out, 256, 0, 0);
if (frames < 0) {
throw EncodeError ("could not run sample-rate converter");
break;
}
- write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels());
+ write_audio (out, frames);
+ }
+
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ delete[] out[i];
}
swr_free (&_swr_context);
}
void
-J2KWAVEncoder::process_audio (uint8_t* data, int size)
+J2KWAVEncoder::process_audio (float** data, int frames)
{
- /* This is a buffer we might use if we are sample-rate converting;
- it will need freeing if so.
- */
- uint8_t* out_buffer = 0;
+ float* resampled[_fs->audio_channels()];
- /* Maybe sample-rate convert */
#if HAVE_SWRESAMPLE
+ /* Maybe sample-rate convert */
if (_swr_context) {
- uint8_t const * in[2] = {
- data,
- 0
- };
+ /* Compute the resampled frames count and add 32 for luck */
+ int const resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
- /* Here's samples per channel */
- int const samples = size / _fs->bytes_per_sample();
-
- /* 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();
-
- /* 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();
- out_buffer = new uint8_t[out_buffer_size_bytes];
-
- uint8_t* out[2] = {
- out_buffer,
- 0
- };
+ /* Make a buffer to put the result in */
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ resampled[i] = new float[resampled_frames];
+ }
/* Resample audio */
- int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, frames);
+ int out_frames = swr_convert (_swr_context, (uint8_t **) resampled, resampled_frames, (uint8_t const **) data, frames);
if (out_frames < 0) {
throw EncodeError ("could not run sample-rate converter");
}
/* And point our variables at the resampled audio */
- data = out_buffer;
- size = out_frames * _fs->audio_channels() * _fs->bytes_per_sample();
+ data = resampled;
+ frames = resampled_frames;
}
#endif
- write_audio (data, size);
+ write_audio (data, frames);
- /* Delete the sample-rate conversion buffer, if it exists */
- delete[] out_buffer;
+#if HAVE_SWRESAMPLE
+ if (_swr_context) {
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ delete[] resampled[i];
+ }
+ }
+#endif
}
void
-J2KWAVEncoder::write_audio (uint8_t* data, int size)
+J2KWAVEncoder::write_audio (float** data, int frames)
{
- /* XXX: we are assuming that the _deinterleave_buffer_size is a multiple
- 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 ((_deinterleave_buffer_size % _fs->bytes_per_sample()) == 0);
-
- /* XXX: this code is very tricksy and it must be possible to make it simpler ... */
-
- /* Number of bytes left to read this time */
- int remaining = size;
- /* Our position in the output buffers, in bytes */
- 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) {
- 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;
- _deinterleave_buffer[to] = data[from];
- }
- }
-
- 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;
- default:
- throw EncodeError ("unknown audio sample format");
- }
- }
-
- position += this_time;
- remaining -= this_time * _fs->audio_channels();
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ sf_write_float (_sound_files[i], data[i], frames);
}
}
projects / dcpomatic.git / blobdiff