#include <iostream>
#include <stdint.h>
+#include <boost/lexical_cast.hpp>
extern "C" {
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersrc.h>
, _video_frame (0)
, _buffer_src_context (0)
, _buffer_sink_context (0)
-#if HAVE_SWRESAMPLE
- , _swr_context (0)
-#endif
, _have_setup_video_filters (false)
, _delay_line (0)
, _delay_in_bytes (0)
void
Decoder::process_begin ()
{
- if (_fs->audio_sample_rate != dcp_audio_sample_rate (_fs->audio_sample_rate)) {
-#if HAVE_SWRESAMPLE
- _swr_context = swr_alloc_set_opts (
- 0,
- audio_channel_layout(),
- audio_sample_format(),
- dcp_audio_sample_rate (_fs->audio_sample_rate),
- audio_channel_layout(),
- audio_sample_format(),
- _fs->audio_sample_rate,
- 0, 0
- );
-
- swr_init (_swr_context);
-#else
- throw DecodeError ("Cannot resample audio as libswresample is not present");
-#endif
- } else {
-#if HAVE_SWRESAMPLE
- _swr_context = 0;
-#endif
- }
-
_delay_in_bytes = _fs->audio_delay * _fs->audio_sample_rate * _fs->audio_channels * _fs->bytes_per_sample() / 1000;
delete _delay_line;
_delay_line = new DelayLine (_delay_in_bytes);
void
Decoder::process_end ()
{
-#if HAVE_SWRESAMPLE
- if (_swr_context) {
-
- int mop = 0;
- while (1) {
- uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels];
- uint8_t* out[1] = {
- buffer
- };
-
- int const frames = swr_convert (_swr_context, out, 256, 0, 0);
-
- if (frames < 0) {
- throw DecodeError ("could not run sample-rate converter");
- }
-
- if (frames == 0) {
- break;
- }
-
- mop += frames;
- int available = _delay_line->feed (buffer, frames * _fs->audio_channels * _fs->bytes_per_sample());
- Audio (buffer, available);
- }
-
- swr_free (&_swr_context);
- }
-#endif
-
if (_delay_in_bytes < 0) {
uint8_t remainder[-_delay_in_bytes];
_delay_line->get_remaining (remainder);
in to get it to the right length.
*/
- int const audio_short_by_frames =
- (decoding_frames() * dcp_audio_sample_rate (_fs->audio_sample_rate) / _fs->frames_per_second)
+ int64_t const audio_short_by_frames =
+ ((int64_t) decoding_frames() * _fs->target_sample_rate() / _fs->frames_per_second)
- _audio_frames_processed;
- int bytes = audio_short_by_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ if (audio_short_by_frames >= 0) {
- int const silence_size = 64 * 1024;
- uint8_t silence[silence_size];
- memset (silence, 0, silence_size);
+ stringstream s;
+ s << "Adding " << audio_short_by_frames << " frames of silence to the end.";
+ _log->log (s.str ());
- while (bytes) {
- int const t = min (bytes, silence_size);
- Audio (silence, t);
- bytes -= t;
+ int64_t bytes = audio_short_by_frames * _fs->audio_channels * _fs->bytes_per_sample();
+
+ int64_t const silence_size = 64 * 1024;
+ uint8_t silence[silence_size];
+ memset (silence, 0, silence_size);
+
+ while (bytes) {
+ int64_t const t = min (bytes, silence_size);
+ Audio (silence, t);
+ bytes -= t;
+ }
}
}
void
Decoder::process_audio (uint8_t* data, int size)
{
- /* Here's samples per channel */
+ /* Samples per channel */
int const samples = size / _fs->bytes_per_sample();
-#if HAVE_SWRESAMPLE
- /* 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;
-#endif
-
/* Maybe apply gain */
if (_fs->audio_gain != 0) {
float const linear_gain = pow (10, _fs->audio_gain / 20);
}
}
- /* This is a buffer we might use if we are sample-rate converting;
- it will need freeing if so.
- */
- uint8_t* out_buffer = 0;
-
- /* Maybe sample-rate convert */
-#if HAVE_SWRESAMPLE
- if (_swr_context) {
-
- uint8_t const * in[2] = {
- data,
- 0
- };
-
- /* Compute the resampled frame count and add 32 for luck */
- int const out_buffer_size_frames = ceil (frames * float (dcp_audio_sample_rate (_fs->audio_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
- };
-
- /* Resample audio */
- int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, frames);
- if (out_frames < 0) {
- throw DecodeError ("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();
- }
-#endif
-
/* Update the number of audio frames we've pushed to the encoder */
_audio_frames_processed += size / (_fs->audio_channels * _fs->bytes_per_sample ());
/* Push into the delay line and then tell the world what we've got */
int available = _delay_line->feed (data, size);
Audio (data, available);
-
- /* Delete the sample-rate conversion buffer, if it exists */
- delete[] out_buffer;
}
/** Called by subclasses to tell the world that some video data is ready.
image->make_black ();
}
+ overlay (image);
+
+ TIMING ("Decoder emits %1", _video_frame);
Video (image, _video_frame);
++_video_frame;
}
<< sample_aspect_ratio_denominator();
int r;
+
if ((r = avfilter_graph_create_filter (&_buffer_src_context, buffer_src, "in", a.str().c_str(), 0, graph)) < 0) {
throw DecodeError ("could not create buffer source");
}
- enum PixelFormat pixel_formats[] = { pixel_format(), PIX_FMT_NONE };
- if (avfilter_graph_create_filter (&_buffer_sink_context, buffer_sink, "out", 0, pixel_formats, graph) < 0) {
+ AVBufferSinkParams* sink_params = av_buffersink_params_alloc ();
+ PixelFormat* pixel_fmts = new PixelFormat[2];
+ pixel_fmts[0] = pixel_format ();
+ pixel_fmts[1] = PIX_FMT_NONE;
+ sink_params->pixel_fmts = pixel_fmts;
+
+ if (avfilter_graph_create_filter (&_buffer_sink_context, buffer_sink, "out", 0, sink_params, graph) < 0) {
throw DecodeError ("could not create buffer sink.");
}
projects / dcpomatic.git / blobdiff