*/
#include "audio_decoder.h"
-#include "stream.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
+#include "i18n.h"
+
+using std::stringstream;
using boost::optional;
using boost::shared_ptr;
-AudioDecoder::AudioDecoder (shared_ptr<Film> f, DecodeOptions o, Job* j)
- : Decoder (f, o, j)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
+ : Decoder (f)
+ , _next_audio (0)
+ , _audio_content (c)
+ , _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f))
+{
+ if (_audio_content->content_audio_frame_rate() != _output_audio_frame_rate) {
+
+ stringstream s;
+ s << String::compose ("Will resample audio from %1 to %2", _audio_content->content_audio_frame_rate(), _output_audio_frame_rate);
+ _film->log()->log (s.str ());
+
+ /* We will be using planar float data when we call the
+ resampler. As far as I can see, the audio channel
+ layout is not necessary for our purposes; it seems
+ only to be used get the number of channels and
+ decide if rematrixing is needed. It won't be, since
+ input and output layouts are the same.
+ */
+
+ _swr_context = swr_alloc_set_opts (
+ 0,
+ av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+ AV_SAMPLE_FMT_FLTP,
+ _output_audio_frame_rate,
+ av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+ AV_SAMPLE_FMT_FLTP,
+ _audio_content->content_audio_frame_rate(),
+ 0, 0
+ );
+
+ swr_init (_swr_context);
+ } else {
+ _swr_context = 0;
+ }
+}
+
+AudioDecoder::~AudioDecoder ()
+{
+ if (_swr_context) {
+ swr_free (&_swr_context);
+ }
+}
+
+
+#if 0
+void
+AudioDecoder::process_end ()
{
+ if (_swr_context) {
+ shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_mapping().dcp_channels(), 256));
+
+ while (1) {
+ int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
+
+ if (frames < 0) {
+ throw EncodeError (_("could not run sample-rate converter"));
+ }
+
+ if (frames == 0) {
+ break;
+ }
+
+ out->set_frames (frames);
+ _writer->write (out);
+ }
+
+ }
}
+#endif
void
-AudioDecoder::set_audio_stream (shared_ptr<AudioStream> s)
+AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
{
- _audio_stream = s;
+ /* XXX: map audio to 5.1 */
+
+ /* Maybe resample */
+ if (_swr_context) {
+
+ /* Compute the resampled frames count and add 32 for luck */
+ int const max_resampled_frames = ceil ((int64_t) data->frames() * _output_audio_frame_rate / _audio_content->content_audio_frame_rate()) + 32;
+
+ shared_ptr<AudioBuffers> resampled (new AudioBuffers (MAX_AUDIO_CHANNELS, max_resampled_frames));
+
+ /* Resample audio */
+ int const resampled_frames = swr_convert (
+ _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames()
+ );
+
+ if (resampled_frames < 0) {
+ throw EncodeError (_("could not run sample-rate converter"));
+ }
+
+ resampled->set_frames (resampled_frames);
+
+ /* And point our variables at the resampled audio */
+ data = resampled;
+ }
+
+ Audio (data, time);
+
+ _next_audio = time + _film->audio_frames_to_time (data->frames());
}
+
+