#include "sndfile_decoder.h"
#include "film.h"
#include "exceptions.h"
+#include "audio_buffers.h"
#include "i18n.h"
using std::vector;
using std::string;
-using std::stringstream;
using std::min;
using std::cout;
using boost::shared_ptr;
-using boost::optional;
-
-/* XXX */
SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
: Decoder (f)
- , AudioDecoder (f)
+ , AudioDecoder (f, c)
+ , _sndfile_content (c)
+ , _deinterleave_buffer (0)
{
- sf_count_t frames;
- SNDFILE* sf = open_file (frames);
- sf_close (sf);
+ _info.format = 0;
+ _sndfile = sf_open (_sndfile_content->path(0).string().c_str(), SFM_READ, &_info);
+ if (!_sndfile) {
+ throw DecodeError (_("could not open audio file for reading"));
+ }
+
+ _done = 0;
+ _remaining = _info.frames;
}
-SNDFILE*
-SndfileDecoder::open_file (sf_count_t & frames)
+SndfileDecoder::~SndfileDecoder ()
{
- frames = 0;
-
- SF_INFO info;
- SNDFILE* s = sf_open (_sndfile_content->file().string().c_str(), SFM_READ, &info);
- if (!s) {
- throw DecodeError (_("could not open external audio file for reading"));
- }
-
- frames = info.frames;
- return s;
+ sf_close (_sndfile);
+ delete[] _deinterleave_buffer;
}
bool
SndfileDecoder::pass ()
{
- sf_count_t frames;
- SNDFILE* sndfile = open_file (frames);
-
+ 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;
-
- shared_ptr<AudioBuffers> audio (new AudioBuffers (_sndfile_content->audio_channels(), block));
- while (frames > 0) {
- sf_count_t const this_time = min (block, frames);
- sf_read_float (sndfile, audio->data(0), this_time);
- audio->set_frames (this_time);
- Audio (audio);
- frames -= this_time;
- }
+ sf_count_t const block = _sndfile_content->content_audio_frame_rate() / 2;
+ sf_count_t const this_time = min (block, _remaining);
- sf_close (sndfile);
+ int const channels = _sndfile_content->audio_channels ();
+
+ shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
+
+ if (_sndfile_content->audio_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 (data, _done * TIME_HZ / audio_frame_rate ());
+ _done += this_time;
+ _remaining -= this_time;
return true;
}
+
+int
+SndfileDecoder::audio_channels () const
+{
+ return _info.channels;
+}
+
+AudioContent::Frame
+SndfileDecoder::audio_length () const
+{
+ return _info.frames;
+}
+
+int
+SndfileDecoder::audio_frame_rate () const
+{
+ return _info.samplerate;
+}
+
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
+{
+ Decoder::seek (t, accurate);
+
+ _done = t * audio_frame_rate() / TIME_HZ;
+ _remaining = _info.frames - _done;
+}