2 Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
4 This program is free software; you can redistribute it and/or modify
5 it under the terms of the GNU General Public License as published by
6 the Free Software Foundation; either version 2 of the License, or
7 (at your option) any later version.
9 This program is distributed in the hope that it will be useful,
10 but WITHOUT ANY WARRANTY; without even the implied warranty of
11 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 GNU General Public License for more details.
14 You should have received a copy of the GNU General Public License
15 along with this program; if not, write to the Free Software
16 Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
20 #include "audio_decoder.h"
21 #include "audio_buffers.h"
22 #include "exceptions.h"
27 using std::stringstream;
31 using boost::optional;
32 using boost::shared_ptr;
34 AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
39 if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
41 shared_ptr<const Film> film = _film.lock ();
45 s << String::compose (
46 "Will resample audio from %1 to %2",
47 _audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate()
50 film->log()->log (s.str ());
52 /* We will be using planar float data when we call the
53 resampler. As far as I can see, the audio channel
54 layout is not necessary for our purposes; it seems
55 only to be used get the number of channels and
56 decide if rematrixing is needed. It won't be, since
57 input and output layouts are the same.
60 _swr_context = swr_alloc_set_opts (
62 av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
64 _audio_content->output_audio_frame_rate(),
65 av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
67 _audio_content->content_audio_frame_rate(),
71 swr_init (_swr_context);
77 AudioDecoder::~AudioDecoder ()
80 swr_free (&_swr_context);
87 AudioDecoder::process_end ()
91 shared_ptr<const Film> film = _film.lock ();
94 shared_ptr<AudioBuffers> out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256));
97 int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
100 throw EncodeError (_("could not run sample-rate converter"));
107 out->set_frames (frames);
108 _writer->write (out);
116 AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
121 /* Compute the resampled frames count and add 32 for luck */
122 int const max_resampled_frames = ceil (
123 (int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
126 shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->channels(), max_resampled_frames));
129 int const resampled_frames = swr_convert (
130 _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames()
133 if (resampled_frames < 0) {
134 throw EncodeError (_("could not run sample-rate converter"));
137 resampled->set_frames (resampled_frames);
139 /* And point our variables at the resampled audio */
143 shared_ptr<const Film> film = _film.lock ();
147 shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
148 dcp_mapped->make_silent ();
149 list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
150 for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
151 dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
154 Audio (dcp_mapped, time);
155 cout << "bumping n.a. by " << data->frames() << " ie " << film->audio_frames_to_time(data->frames()) << "\n";
156 _next_audio = time + film->audio_frames_to_time (data->frames());
160 AudioDecoder::audio_done () const
162 shared_ptr<const Film> film = _film.lock ();
165 return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);