#include <iostream>
#include <stdint.h>
+#include <boost/lexical_cast.hpp>
extern "C" {
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersrc.h>
#include "film.h"
#include "format.h"
#include "job.h"
-#include "film_state.h"
#include "options.h"
#include "exceptions.h"
#include "image.h"
#include "filter.h"
#include "delay_line.h"
#include "ffmpeg_compatibility.h"
+#include "subtitle.h"
using namespace std;
using namespace boost;
-/** @param s FilmState of the Film.
+/** @param f Film.
* @param o Options.
* @param j Job that we are running within, or 0
- * @param l Log to use.
* @param minimal true to do the bare minimum of work; just run through the content. Useful for acquiring
* accurate frame counts as quickly as possible. This generates no video or audio output.
* @param ignore_length Ignore the content's claimed length when computing progress.
*/
-Decoder::Decoder (boost::shared_ptr<const FilmState> s, boost::shared_ptr<const Options> o, Job* j, Log* l, bool minimal, bool ignore_length)
- : _fs (s)
+Decoder::Decoder (boost::shared_ptr<Film> f, boost::shared_ptr<const Options> o, Job* j, bool minimal, bool ignore_length)
+ : _film (f)
, _opt (o)
, _job (j)
- , _log (l)
, _minimal (minimal)
, _ignore_length (ignore_length)
, _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)
, _audio_frames_processed (0)
{
- if (_opt->decode_video_frequency != 0 && _fs->length == 0) {
+ if (_opt->decode_video_frequency != 0 && _film->length() == 0) {
throw DecodeError ("cannot do a partial decode if length == 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;
+ _delay_in_bytes = _film->audio_delay() * _film->audio_sample_rate() * _film->audio_channels() * bytes_per_audio_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);
- _audio_frames_processed += _delay_in_bytes / (_fs->audio_channels * _fs->bytes_per_sample());
- Audio (remainder, _delay_in_bytes);
+ _audio_frames_processed += _delay_in_bytes / (_film->audio_channels() * bytes_per_audio_sample());
+ emit_audio (remainder, -_delay_in_bytes);
}
/* If we cut the decode off, the audio may be short; push some silence
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)
- - _audio_frames_processed;
+ int64_t const video_length_in_audio_frames = ((int64_t) _film->dcp_length() * _film->audio_sample_rate() / _film->frames_per_second());
+ int64_t const audio_short_by_frames = video_length_in_audio_frames - _audio_frames_processed;
+
+ _log->log (
+ String::compose ("DCP length is %1 (%2 audio frames); %3 frames of audio processed.",
+ _film->dcp_length(),
+ video_length_in_audio_frames,
+ _audio_frames_processed)
+ );
+
+ if (audio_short_by_frames >= 0 && _opt->decode_audio) {
- int bytes = audio_short_by_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ _log->log (String::compose ("DCP length is %1; %2 frames of audio processed.", _film->dcp_length(), _audio_frames_processed));
+ _log->log (String::compose ("Adding %1 frames of silence to the end.", audio_short_by_frames));
- int const silence_size = 64 * 1024;
- uint8_t silence[silence_size];
- memset (silence, 0, silence_size);
+ /* XXX: this is slightly questionable; does memset () give silence with all
+ sample formats?
+ */
- while (bytes) {
- int const t = min (bytes, silence_size);
- Audio (silence, t);
- bytes -= t;
+ int64_t bytes = audio_short_by_frames * _film->audio_channels() * bytes_per_audio_sample();
+
+ int64_t const silence_size = 16 * 1024 * _film->audio_channels() * bytes_per_audio_sample();
+ uint8_t silence[silence_size];
+ memset (silence, 0, silence_size);
+
+ while (bytes) {
+ int64_t const t = min (bytes, silence_size);
+ emit_audio (silence, t);
+ bytes -= t;
+ }
}
}
while (pass () == false) {
if (_job && !_ignore_length) {
- _job->set_progress (float (_video_frame) / decoding_frames ());
+ _job->set_progress (float (_video_frame) / _film->dcp_length());
}
}
process_end ();
}
-/** @return Number of frames that we will be decoding */
-int
-Decoder::decoding_frames () const
-{
- if (_opt->num_frames > 0) {
- return _opt->num_frames;
- }
-
- return _fs->length;
-}
-
/** Run one pass. This may or may not generate any actual video / audio data;
* some decoders may require several passes to generate a single frame.
* @return true if we have finished processing all data; otherwise false.
_have_setup_video_filters = true;
}
- if (_opt->num_frames != 0 && _video_frame >= _opt->num_frames) {
+ if (!_ignore_length && _video_frame >= _film->dcp_length()) {
return true;
}
}
/** Called by subclasses to tell the world that some audio data is ready
- * @param data Interleaved audio data, in FilmState::audio_sample_format.
+ * @param data Audio data, in Film::audio_sample_format.
* @param size Number of bytes of data.
*/
void
Decoder::process_audio (uint8_t* data, int size)
{
- /* Here's samples per channel */
- int const samples = size / _fs->bytes_per_sample();
+ /* Push into the delay line */
+ size = _delay_line->feed (data, size);
-#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
+ emit_audio (data, size);
+}
- /* Maybe apply gain */
- if (_fs->audio_gain != 0) {
- float const linear_gain = pow (10, _fs->audio_gain / 20);
- uint8_t* p = data;
- switch (_fs->audio_sample_format) {
- case AV_SAMPLE_FMT_S16:
- for (int i = 0; i < samples; ++i) {
- /* XXX: assumes little-endian; also we should probably be dithering here */
-
- /* unsigned sample */
- int const ou = p[0] | (p[1] << 8);
-
- /* signed sample */
- int const os = ou >= 0x8000 ? (- 0x10000 + ou) : ou;
-
- /* signed sample with altered gain */
- int const gs = int (os * linear_gain);
-
- /* unsigned sample with altered gain */
- int const gu = gs > 0 ? gs : (0x10000 + gs);
-
- /* write it back */
- p[0] = gu & 0xff;
- p[1] = (gu & 0xff00) >> 8;
- p += 2;
+void
+Decoder::emit_audio (uint8_t* data, int size)
+{
+ /* Deinterleave and convert to float */
+
+ assert ((size % (bytes_per_audio_sample() * _film->audio_channels())) == 0);
+
+ int const total_samples = size / bytes_per_audio_sample();
+ int const frames = total_samples / _film->audio_channels();
+ shared_ptr<AudioBuffers> audio (new AudioBuffers (_film->audio_channels(), frames));
+
+ switch (audio_sample_format()) {
+ case AV_SAMPLE_FMT_S16:
+ {
+ int16_t* p = (int16_t *) data;
+ int sample = 0;
+ int channel = 0;
+ for (int i = 0; i < total_samples; ++i) {
+ audio->data(channel)[sample] = float(*p++) / (1 << 15);
+
+ ++channel;
+ if (channel == _film->audio_channels()) {
+ channel = 0;
+ ++sample;
+ }
+ }
+ }
+ break;
+
+ case AV_SAMPLE_FMT_S32:
+ {
+ int32_t* p = (int32_t *) data;
+ int sample = 0;
+ int channel = 0;
+ for (int i = 0; i < total_samples; ++i) {
+ audio->data(channel)[sample] = float(*p++) / (1 << 31);
+
+ ++channel;
+ if (channel == _film->audio_channels()) {
+ channel = 0;
+ ++sample;
}
- break;
- default:
- assert (false);
}
}
- /* 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");
+ case AV_SAMPLE_FMT_FLTP:
+ {
+ float* p = reinterpret_cast<float*> (data);
+ for (int i = 0; i < _film->audio_channels(); ++i) {
+ memcpy (audio->data(i), p, frames * sizeof(float));
+ p += frames;
}
+ }
+ break;
- /* And point our variables at the resampled audio */
- data = out_buffer;
- size = out_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ default:
+ assert (false);
+ }
+
+ /* Maybe apply gain */
+ if (_film->audio_gain() != 0) {
+ float const linear_gain = pow (10, _film->audio_gain() / 20);
+ for (int i = 0; i < _film->audio_channels(); ++i) {
+ for (int j = 0; j < frames; ++j) {
+ audio->data(i)[j] *= linear_gain;
+ }
+ }
}
-#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);
+ /* Update the number of audio frames we've pushed to the encoder */
+ _audio_frames_processed += audio->frames ();
- /* Delete the sample-rate conversion buffer, if it exists */
- delete[] out_buffer;
+ Audio (audio);
}
/** Called by subclasses to tell the world that some video data is ready.
return;
}
- /* Use FilmState::length here as our one may be wrong */
+ /* Use Film::length here as our one may be wrong */
int gap = 0;
if (_opt->decode_video_frequency != 0) {
- gap = _fs->length / _opt->decode_video_frequency;
+ gap = _film->length() / _opt->decode_video_frequency;
}
if (_opt->decode_video_frequency != 0 && gap != 0 && (_video_frame % gap) != 0) {
image->make_black ();
}
- Video (image, _video_frame);
+ shared_ptr<Subtitle> sub;
+ if (_timed_subtitle && _timed_subtitle->displayed_at (double (last_video_frame()) / _film->frames_per_second())) {
+ sub = _timed_subtitle->subtitle ();
+ }
+
+ TIMING ("Decoder emits %1", _video_frame);
+ Video (image, _video_frame, sub);
++_video_frame;
}
}
Size size_after_crop;
if (_opt->apply_crop) {
- size_after_crop = _fs->cropped_size (native_size ());
- fs << crop_string (Position (_fs->crop.left, _fs->crop.top), size_after_crop);
+ size_after_crop = _film->cropped_size (native_size ());
+ fs << crop_string (Position (_film->crop().left, _film->crop().top), size_after_crop);
} else {
size_after_crop = native_size ();
fs << crop_string (Position (0, 0), size_after_crop);
}
- string filters = Filter::ffmpeg_strings (_fs->filters).first;
+ string filters = Filter::ffmpeg_strings (_film->filters()).first;
if (!filters.empty ()) {
filters += ",";
}
<< 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.");
}
/* XXX: leaking `inputs' / `outputs' ? */
}
+void
+Decoder::process_subtitle (shared_ptr<TimedSubtitle> s)
+{
+ _timed_subtitle = s;
+
+ if (_timed_subtitle && _opt->apply_crop) {
+ Position const p = _timed_subtitle->subtitle()->position ();
+ _timed_subtitle->subtitle()->set_position (Position (p.x - _film->crop().left, p.y - _film->crop().top));
+ }
+}
+
+
+int
+Decoder::bytes_per_audio_sample () const
+{
+ return av_get_bytes_per_sample (audio_sample_format ());
+}
projects / dcpomatic.git / blobdiff