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/*
Copyright (C) 2013 Carl Hetherington <cth@carlh.net>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
extern "C" {
#include "libavutil/channel_layout.h"
}
#include "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "compose.hpp"
#include "i18n.h"
using std::cout;
using std::pair;
using std::make_pair;
using boost::shared_ptr;
Resampler::Resampler (int in, int out, int channels)
: _in_rate (in)
, _out_rate (out)
, _channels (channels)
{
/* 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 (_channels),
AV_SAMPLE_FMT_FLTP,
_out_rate,
av_get_default_channel_layout (_channels),
AV_SAMPLE_FMT_FLTP,
_in_rate,
0, 0
);
swr_init (_swr_context);
}
Resampler::~Resampler ()
{
swr_free (&_swr_context);
}
pair<shared_ptr<const AudioBuffers>, AudioContent::Frame>
Resampler::run (shared_ptr<const AudioBuffers> in, AudioContent::Frame frame)
{
AudioContent::Frame const resamp_time = swr_next_pts (_swr_context, frame * _out_rate) / _in_rate;
/* Compute the resampled frames count and add 32 for luck */
int const max_resampled_frames = ceil ((double) in->frames() * _out_rate / _in_rate) + 32;
shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, max_resampled_frames));
int const resampled_frames = swr_convert (
_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) in->data(), in->frames()
);
if (resampled_frames < 0) {
char buf[256];
av_strerror (resampled_frames, buf, sizeof(buf));
throw EncodeError (String::compose (_("could not run sample-rate converter for %1 samples (%2) (%3)"), in->frames(), resampled_frames, buf));
}
resampled->set_frames (resampled_frames);
return make_pair (resampled, resamp_time);
}
shared_ptr<const AudioBuffers>
Resampler::flush ()
{
shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
int out_offset = 0;
int64_t const pass_size = 256;
shared_ptr<AudioBuffers> pass (new AudioBuffers (_channels, 256));
while (1) {
int const frames = swr_convert (_swr_context, (uint8_t **) pass->data(), pass_size, 0, 0);
if (frames < 0) {
throw EncodeError (_("could not run sample-rate converter"));
}
if (frames == 0) {
break;
}
out->ensure_size (out_offset + frames);
out->copy_from (pass.get(), frames, 0, out_offset);
out_offset += frames;
out->set_frames (out_offset);
}
return out;
}
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