void
ABTranscoder::go ()
{
- _encoder->process_begin ();
+ _encoder->process_begin (_da->audio_channel_layout(), _da->audio_sample_format());
_da->process_begin ();
_db->process_begin ();
, _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);
*/
int64_t const audio_short_by_frames =
- ((int64_t) decoding_frames() * dcp_audio_sample_rate (_fs->audio_sample_rate) / _fs->frames_per_second)
+ ((int64_t) decoding_frames() * _fs->audio_sample_rate / _fs->frames_per_second)
- _audio_frames_processed;
if (audio_short_by_frames >= 0) {
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.
#include <stdint.h>
#include <boost/shared_ptr.hpp>
#include <sigc++/sigc++.h>
-#ifdef HAVE_SWRESAMPLE
-extern "C" {
-#include <libswresample/swresample.h>
-}
-#endif
#include "util.h"
class Job;
AVFilterContext* _buffer_src_context;
AVFilterContext* _buffer_sink_context;
-#if HAVE_SWRESAMPLE
- SwrContext* _swr_context;
-#endif
-
bool _have_setup_video_filters;
DelayLine* _delay_line;
int _delay_in_bytes;
#include <boost/thread/mutex.hpp>
#include <list>
#include <stdint.h>
+extern "C" {
+#include <libavutil/samplefmt.h>
+}
class FilmState;
class Options;
Encoder (boost::shared_ptr<const FilmState> s, boost::shared_ptr<const Options> o, Log* l);
/** Called to indicate that a processing run is about to begin */
- virtual void process_begin () = 0;
+ virtual void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) = 0;
/** Called with a frame of video.
* @param i Video frame image.
public:
J2KStillEncoder (boost::shared_ptr<const FilmState>, boost::shared_ptr<const Options>, Log *);
- void process_begin () {}
+ void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
void process_video (boost::shared_ptr<Image>, int);
void process_audio (uint8_t *, int) {}
void process_end () {}
J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const FilmState> s, shared_ptr<const Options> o, Log* l)
: Encoder (s, o, l)
+#ifdef HAVE_SWRESAMPLE
+ , _swr_context (0)
+#endif
, _deinterleave_buffer_size (8192)
, _deinterleave_buffer (0)
, _process_end (false)
}
void
-J2KWAVEncoder::process_begin ()
+J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format)
{
+ if ((_fs->audio_sample_rate != dcp_audio_sample_rate (_fs->audio_sample_rate)) || (rint (_fs->frames_per_second) != _fs->frames_per_second)) {
+#ifdef HAVE_SWRESAMPLE
+ _swr_context = swr_alloc_set_opts (
+ 0,
+ audio_channel_layout,
+ audio_sample_format,
+ target_sample_rate(),
+ audio_channel_layout,
+ audio_sample_format,
+ _fs->audio_sample_rate,
+ 0, 0
+ );
+
+ swr_init (_swr_context);
+#else
+ throw EncodeError ("Cannot resample audio as libswresample is not present");
+#endif
+ } else {
+#ifdef HAVE_SWRESAMPLE
+ _swr_context = 0;
+#endif
+ }
+
for (int i = 0; i < Config::instance()->num_local_encoding_threads (); ++i) {
_worker_threads.push_back (new boost::thread (boost::bind (&J2KWAVEncoder::encoder_thread, this, (ServerDescription *) 0)));
}
_log->log (s.str ());
}
}
+
+#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 EncodeError ("could not run sample-rate converter");
+ }
+
+ if (frames == 0) {
+ break;
+ }
+
+ mop += frames;
+ write_audio (buffer, frames);
+ }
+
+ swr_free (&_swr_context);
+ }
+#endif
close_sound_files ();
}
void
-J2KWAVEncoder::process_audio (uint8_t* data, int data_size)
+J2KWAVEncoder::process_audio (uint8_t* data, int size)
+{
+ /* 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
+ };
+
+ /* Here's samples per channel */
+ int const samples = size / _fs->bytes_per_sample();
+
+ /* 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;
+
+ /* Compute the resampled frame count and add 32 for luck */
+ int const out_buffer_size_frames = ceil (frames * target_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, size);
+ if (out_frames < 0) {
+ throw EncodeError ("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
+
+ write_audio (data, size);
+
+ /* Delete the sample-rate conversion buffer, if it exists */
+ delete[] out_buffer;
+}
+
+void
+J2KWAVEncoder::write_audio (uint8_t* data, int size)
{
/* Size of a sample in bytes */
int const sample_size = 2;
-
+
/* XXX: we are assuming that sample_size is right, the _deinterleave_buffer_size is a multiple
of the sample size and that data_size is a multiple of _fs->audio_channels * sample_size.
*/
/* XXX: this code is very tricksy and it must be possible to make it simpler ... */
/* Number of bytes left to read this time */
- int remaining = data_size;
+ int remaining = size;
/* Our position in the output buffers, in bytes */
int position = 0;
while (remaining > 0) {
sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / sample_size);
break;
default:
- throw DecodeError ("unknown audio sample format");
+ throw EncodeError ("unknown audio sample format");
}
}
remaining -= this_time * _fs->audio_channels;
}
}
+
+int
+J2KWAVEncoder::target_sample_rate () const
+{
+ double t = dcp_audio_sample_rate (_fs->audio_sample_rate);
+ if (rint (_fs->frames_per_second) != _fs->frames_per_second) {
+ if (_fs->frames_per_second == 23.976) {
+ /* 24fps drop-frame ie 24 * 1000 / 1001 frames per second;
+ hence we need to resample the audio to dcp_audio_sample_rate * 1000 / 1001
+ so that when we play it back at dcp_audio_sample_rate it is sped up
+ by the same amount that the video is
+ */
+ t *= double(1000) / 1001;
+ } else {
+ throw EncodeError ("unknown fractional frame rate");
+ }
+ }
+
+ return rint (t);
+}
#include <boost/thread/condition.hpp>
#include <boost/thread/mutex.hpp>
#include <boost/thread.hpp>
+#ifdef HAVE_SWRESAMPLE
+extern "C" {
+#include <libswresample/swresample.h>
+}
+#endif
#include <sndfile.h>
#include "encoder.h"
J2KWAVEncoder (boost::shared_ptr<const FilmState>, boost::shared_ptr<const Options>, Log *);
~J2KWAVEncoder ();
- void process_begin ();
+ void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format);
void process_video (boost::shared_ptr<Image>, int);
void process_audio (uint8_t *, int);
void process_end ();
-private:
+private:
+ int target_sample_rate () const;
+
+ void write_audio (uint8_t* data, int size);
void encoder_thread (ServerDescription *);
void close_sound_files ();
void terminate_worker_threads ();
+#if HAVE_SWRESAMPLE
+ SwrContext* _swr_context;
+#endif
+
std::vector<SNDFILE*> _sound_files;
int _deinterleave_buffer_size;
uint8_t* _deinterleave_buffer;
public:
TIFFEncoder (boost::shared_ptr<const FilmState> s, boost::shared_ptr<const Options> o, Log* l);
- void process_begin () {}
+ void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
void process_video (boost::shared_ptr<Image>, int);
void process_audio (uint8_t *, int) {}
void process_end () {}
void
Transcoder::go ()
{
- _encoder->process_begin ();
+ _encoder->process_begin (_decoder->audio_channel_layout(), _decoder->audio_sample_format());
try {
_decoder->go ();
} catch (...) {
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