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);
+ emit_audio (remainder, _delay_in_bytes);
}
/* If we cut the decode off, the audio may be short; push some silence
while (bytes) {
int64_t const t = min (bytes, silence_size);
- Audio (silence, t);
+ emit_audio (silence, t);
bytes -= t;
}
}
}
/** 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 FilmState::audio_sample_format.
* @param size Number of bytes of data.
*/
void
Decoder::process_audio (uint8_t* data, int size)
{
- /* Samples per channel */
- int const samples = size / _fs->bytes_per_sample();
+ /* Push into the delay line */
+ size = _delay_line->feed (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 */
+ emit_audio (data, size);
+}
- /* unsigned sample */
- int const ou = p[0] | (p[1] << 8);
+void
+Decoder::emit_audio (uint8_t* data, int size)
+{
+ /* Deinterleave and convert to float */
+
+ float* samples[_fs->audio_channels()];
+ int const total_samples = size / _fs->bytes_per_sample();
+ int const frames = total_samples / _fs->audio_channels();
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ samples[i] = new float[frames];
+ }
- /* signed sample */
- int const os = ou >= 0x8000 ? (- 0x10000 + ou) : ou;
+ switch (_fs->audio_sample_format()) {
+ case AV_SAMPLE_FMT_S16:
+ {
+ uint8_t* p = data;
+ int sample = 0;
+ int channel = 0;
+ for (int i = 0; i < total_samples; ++i) {
+ /* unsigned sample */
+ int const ou = p[0] | (p[1] << 8);
+ /* signed sample */
+ int const os = ou >= 0x8000 ? (- 0x10000 + ou) : ou;
+ /* float sample */
+ samples[channel][sample] = float(os) / 0x8000;
+
+ cout << samples[channel][sample] << " from s16\n";
+
+ ++channel;
+ if (channel == _fs->audio_channels()) {
+ channel = 0;
+ ++sample;
+ }
- /* signed sample with altered gain */
- int const gs = int (os * linear_gain);
+ p += 2;
+ }
+ }
+ break;
+
+ case AV_SAMPLE_FMT_FLTP:
+ {
+ float* p = reinterpret_cast<float*> (data);
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ for (int j = 0; j < frames; ++j) {
+ samples[i][j] = *p++;
+ cout << samples[i][j] << " from float.\n";
+ ++p;
+ }
+ }
+ }
+ break;
- /* unsigned sample with altered gain */
- int const gu = gs > 0 ? gs : (0x10000 + gs);
+ default:
+ assert (false);
+ }
- /* write it back */
- p[0] = gu & 0xff;
- p[1] = (gu & 0xff00) >> 8;
- p += 2;
+ /* Maybe apply gain */
+ if (_fs->audio_gain() != 0) {
+ float const linear_gain = pow (10, _fs->audio_gain() / 20);
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ for (int j = 0; j < frames; ++j) {
+ samples[i][j] *= linear_gain;
}
- break;
- default:
- assert (false);
}
}
/* Update the number of audio frames we've pushed to the encoder */
- _audio_frames_processed += size / (_fs->audio_channels() * _fs->bytes_per_sample());
+ _audio_frames_processed += frames;
- /* Push into the delay line and then tell the world what we've got */
- int available = _delay_line->feed (data, size);
- Audio (data, available);
+ Audio (samples, frames);
+
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ delete[] samples[i];
+ }
}
/** Called by subclasses to tell the world that some video data is ready.
sigc::signal<void, boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle> > Video;
/** Emitted when some audio data is ready.
- * First parameter is the interleaved sample data, format is given in the FilmState.
- * Second parameter is the size of the data.
+ * First parameter is an array of pointers to deinterleaved, floating point sample data for each channel.
+ * Second parameter is the size of the data in frames (ie samples on each channel).
*/
- sigc::signal<void, uint8_t *, int> Audio;
+ sigc::signal<void, float**, int> Audio;
protected:
/** perform a single pass at our content */
private:
void setup_video_filters ();
+ void emit_audio (uint8_t* data, int size);
/** last video frame to be processed */
int _video_frame;
virtual void process_video (boost::shared_ptr<Image> i, int f, boost::shared_ptr<Subtitle> s) = 0;
/** Called with some audio data.
- * @param d Data.
- * @param s Size of data (in bytes)
+ * @param d Array of pointers to floating point sample data for each channel.
+ * @param s Number of frames (ie number of samples in each channel)
*/
- virtual void process_audio (uint8_t* d, int s) = 0;
+ virtual void process_audio (float** d, int s) = 0;
/** Called when a processing run has finished */
virtual void process_end () = 0;
int
FilmState::bytes_per_sample () const
{
- switch (_audio_sample_format) {
- case AV_SAMPLE_FMT_S16:
- return 2;
- default:
- return 0;
- }
-
- return 0;
+ return av_get_bytes_per_sample (_audio_sample_format);
}
int
* @param p Pixel format.
* @param s Size in pixels.
*/
-SimpleImage::SimpleImage (PixelFormat p, Size s, function<int (int)> rounder)
+SimpleImage::SimpleImage (AVPixelFormat p, Size s, function<int (int)> rounder)
: Image (p)
, _size (s)
{
return _size;
}
-AlignedImage::AlignedImage (PixelFormat f, Size s)
+AlignedImage::AlignedImage (AVPixelFormat f, Size s)
: SimpleImage (f, s, boost::bind (round_up, _1, 32))
{
}
-CompactImage::CompactImage (PixelFormat f, Size s)
+CompactImage::CompactImage (AVPixelFormat f, Size s)
: SimpleImage (f, s, boost::bind (round_up, _1, 1))
{
}
}
-FilterBufferImage::FilterBufferImage (PixelFormat p, AVFilterBufferRef* b)
+FilterBufferImage::FilterBufferImage (AVPixelFormat p, AVFilterBufferRef* b)
: Image (p)
, _buffer (b)
{
class Image
{
public:
- Image (PixelFormat p)
+ Image (AVPixelFormat p)
: _pixel_format (p)
{}
void read_from_socket (boost::shared_ptr<Socket>);
void write_to_socket (boost::shared_ptr<Socket>) const;
- PixelFormat pixel_format () const {
+ AVPixelFormat pixel_format () const {
return _pixel_format;
}
private:
- PixelFormat _pixel_format; ///< FFmpeg's way of describing the pixel format of this Image
+ AVPixelFormat _pixel_format; ///< FFmpeg's way of describing the pixel format of this Image
};
/** @class FilterBufferImage
class FilterBufferImage : public Image
{
public:
- FilterBufferImage (PixelFormat, AVFilterBufferRef *);
+ FilterBufferImage (AVPixelFormat, AVFilterBufferRef *);
~FilterBufferImage ();
uint8_t ** data () const;
class SimpleImage : public Image
{
public:
- SimpleImage (PixelFormat, Size, boost::function<int (int)> rounder);
+ SimpleImage (AVPixelFormat, Size, boost::function<int (int)> rounder);
~SimpleImage ();
uint8_t ** data () const;
class AlignedImage : public SimpleImage
{
public:
- AlignedImage (PixelFormat, Size);
+ AlignedImage (AVPixelFormat, Size);
};
class CompactImage : public SimpleImage
{
public:
- CompactImage (PixelFormat, Size);
+ CompactImage (AVPixelFormat, Size);
CompactImage (boost::shared_ptr<Image>);
};
void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
- void process_audio (uint8_t *, int) {}
+ void process_audio (float**, int) {}
void process_end () {}
};
void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format) {}
void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
- void process_audio (uint8_t *, int) {}
+ void process_audio (float**, int) {}
void process_end () {}
};
#ifdef HAVE_SWRESAMPLE
, _swr_context (0)
#endif
- , _deinterleave_buffer_size (8192)
- , _deinterleave_buffer (0)
, _process_end (false)
{
/* Create sound output files with .tmp suffixes; we will rename
}
_sound_files.push_back (f);
}
-
- /* Create buffer for deinterleaving audio */
- _deinterleave_buffer = new uint8_t[_deinterleave_buffer_size];
}
J2KWAVEncoder::~J2KWAVEncoder ()
{
terminate_worker_threads ();
- delete[] _deinterleave_buffer;
close_sound_files ();
}
stringstream s;
s << "Will resample audio from " << _fs->audio_sample_rate() << " to " << _fs->target_sample_rate();
_log->log (s.str ());
-
+
+ /* We will be using planar float data when we call the resampler */
_swr_context = swr_alloc_set_opts (
0,
audio_channel_layout,
- audio_sample_format,
+ AV_SAMPLE_FMT_FLTP,
_fs->target_sample_rate(),
audio_channel_layout,
- audio_sample_format,
+ AV_SAMPLE_FMT_FLTP,
_fs->audio_sample_rate(),
0, 0
);
#if HAVE_SWRESAMPLE
if (_swr_context) {
+ float* out[_fs->audio_channels()];
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ out[i] = new float[256];
+ }
+
while (1) {
- uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels()];
- uint8_t* out[2] = {
- buffer,
- 0
- };
-
- int const frames = swr_convert (_swr_context, out, 256, 0, 0);
+ int const frames = swr_convert (_swr_context, (uint8_t **) out, 256, 0, 0);
if (frames < 0) {
throw EncodeError ("could not run sample-rate converter");
break;
}
- write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels());
+ write_audio (out, frames);
+ }
+
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ delete[] out[i];
}
swr_free (&_swr_context);
}
void
-J2KWAVEncoder::process_audio (uint8_t* data, int size)
+J2KWAVEncoder::process_audio (float** data, int frames)
{
- /* This is a buffer we might use if we are sample-rate converting;
- it will need freeing if so.
- */
- uint8_t* out_buffer = 0;
+ float* resampled[_fs->audio_channels()];
- /* Maybe sample-rate convert */
#if HAVE_SWRESAMPLE
+ /* Maybe sample-rate convert */
if (_swr_context) {
- uint8_t const * in[2] = {
- data,
- 0
- };
+ /* Compute the resampled frames count and add 32 for luck */
+ int const resampled_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate()) + 32;
- /* 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 * _fs->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
- };
+ /* Make a buffer to put the result in */
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ resampled[i] = new float[resampled_frames];
+ }
/* Resample audio */
- int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, frames);
+ int out_frames = swr_convert (_swr_context, (uint8_t **) resampled, resampled_frames, (uint8_t const **) data, frames);
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();
+ data = resampled;
+ frames = resampled_frames;
}
#endif
- write_audio (data, size);
+ write_audio (data, frames);
- /* Delete the sample-rate conversion buffer, if it exists */
- delete[] out_buffer;
+#if HAVE_SWRESAMPLE
+ if (_swr_context) {
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ delete[] resampled[i];
+ }
+ }
+#endif
}
void
-J2KWAVEncoder::write_audio (uint8_t* data, int size)
+J2KWAVEncoder::write_audio (float** data, int frames)
{
- /* XXX: we are assuming that the _deinterleave_buffer_size is a multiple
- of the sample size and that size is a multiple of _fs->audio_channels * sample_size.
- */
-
- assert ((size % (_fs->audio_channels() * _fs->bytes_per_sample())) == 0);
- assert ((_deinterleave_buffer_size % _fs->bytes_per_sample()) == 0);
-
- /* 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 = size;
- /* Our position in the output buffers, in bytes */
- int position = 0;
- while (remaining > 0) {
- /* How many bytes of the deinterleaved data to do this time */
- int this_time = min (remaining / _fs->audio_channels(), _deinterleave_buffer_size);
- for (int i = 0; i < _fs->audio_channels(); ++i) {
- for (int j = 0; j < this_time; j += _fs->bytes_per_sample()) {
- for (int k = 0; k < _fs->bytes_per_sample(); ++k) {
- int const to = j + k;
- int const from = position + (i * _fs->bytes_per_sample()) + (j * _fs->audio_channels()) + k;
- _deinterleave_buffer[to] = data[from];
- }
- }
-
- switch (_fs->audio_sample_format()) {
- case AV_SAMPLE_FMT_S16:
- sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / _fs->bytes_per_sample());
- break;
- default:
- throw EncodeError ("unknown audio sample format");
- }
- }
-
- position += this_time;
- remaining -= this_time * _fs->audio_channels();
+ for (int i = 0; i < _fs->audio_channels(); ++i) {
+ sf_write_float (_sound_files[i], data[i], frames);
}
}
void process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format);
void process_video (boost::shared_ptr<Image>, int, boost::shared_ptr<Subtitle>);
- void process_audio (uint8_t *, int);
+ void process_audio (float**, int);
void process_end ();
private:
- void write_audio (uint8_t* data, int size);
+ void write_audio (float** data, int frames);
void encoder_thread (ServerDescription *);
void close_sound_files ();
void terminate_worker_threads ();
#endif
std::vector<SNDFILE*> _sound_files;
- int _deinterleave_buffer_size;
- uint8_t* _deinterleave_buffer;
bool _process_end;
std::list<boost::shared_ptr<DCPVideoFrame> > _queue;
case AV_SAMPLE_FMT_S16:
return "S16";
default:
- break;
+ assert (false);
}
-
- return "Unknown";
}
/** @param s String representation of a sample format, as returned from audio_sample_format_to_string().
return AV_SAMPLE_FMT_S16;
}
- return AV_SAMPLE_FMT_NONE;
+ assert (false);
}
/** @return Version of vobcopy that is on the path (and hence that we will use) */
projects / dcpomatic.git / commitdiff