*/
#include "audio_decoder.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
+#include "i18n.h"
+
+using std::stringstream;
+using std::list;
+using std::pair;
+using std::cout;
using boost::optional;
using boost::shared_ptr;
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
: Decoder (f)
+ , _next_audio (0)
+ , _audio_content (c)
+{
+ if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
+
+ shared_ptr<const Film> film = _film.lock ();
+ assert (film);
+
+ stringstream s;
+ s << String::compose (
+ "Will resample audio from %1 to %2",
+ _audio_content->content_audio_frame_rate(), _audio_content->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,
+ _audio_content->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<const Film> film = _film.lock ();
+ assert (film);
+
+ 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::audio (shared_ptr<const AudioBuffers> data, Time time)
+{
+ /* 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() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
+ ) + 32;
+
+ shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->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;
+ }
+
+ shared_ptr<const Film> film = _film.lock ();
+ assert (film);
+
+ /* Remap channels */
+ shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
+ dcp_mapped->make_silent ();
+ list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
+ for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
+ dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
+ }
+
+ Audio (dcp_mapped, time);
+ cout << "bumping n.a. by " << data->frames() << " ie " << film->audio_frames_to_time(data->frames()) << "\n";
+ _next_audio = time + film->audio_frames_to_time (data->frames());
+}
+
+bool
+AudioDecoder::audio_done () const
+{
+ shared_ptr<const Film> film = _film.lock ();
+ assert (film);
+
+ return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);
}
+