#include "audio_buffers.h"
#include "exceptions.h"
#include "log.h"
+#include "resampler.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, shared_ptr<const AudioContent> c)
- : Decoder (f)
- , _audio_content (c)
- , _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f))
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+ : Decoder (film)
+ , _audio_content (content)
+ , _audio_position (0)
{
- 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 ()
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
{
- 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);
- }
-
- }
+ Audio (data, frame);
+ _audio_position = frame + data->frames ();
}
-#endif
-void
-AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+/** This is a bit odd, but necessary when we have (e.g.) FFmpegDecoders with no audio.
+ * The player needs to know that there is no audio otherwise it will keep trying to
+ * pass() the decoder to get it to emit audio.
+ */
+bool
+AudioDecoder::has_audio () const
{
- /* XXX: map audio to 5.1 */
-
- /* Maybe sample-rate convert */
- 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);
+ return _audio_content->audio_channels () > 0;
}
-
-