2 Copyright (C) 2015 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
6 DCP-o-matic is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
11 DCP-o-matic is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
16 You should have received a copy of the GNU General Public License
17 along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
21 #include "audio_filter_graph.h"
22 #include "audio_buffers.h"
23 #include "compose.hpp"
25 #include <libavfilter/buffersink.h>
26 #include <libavfilter/buffersrc.h>
27 #include <libavutil/channel_layout.h>
33 using boost::shared_ptr;
35 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
36 : _sample_rate (sample_rate)
37 , _channels (channels)
39 /* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
40 so we need to tell it we're using 16 channels if we are using more than 8.
43 _channel_layout = av_get_default_channel_layout (16);
45 _channel_layout = av_get_default_channel_layout (_channels);
48 _in_frame = av_frame_alloc ();
51 AudioFilterGraph::~AudioFilterGraph()
53 av_frame_free (&_in_frame);
57 AudioFilterGraph::src_parameters () const
60 av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
64 buffer, sizeof(buffer), "time_base=1/1:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
65 _sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), layout
72 AudioFilterGraph::sink_parameters () const
74 AVABufferSinkParams* sink_params = av_abuffersink_params_alloc ();
76 AVSampleFormat* sample_fmts = new AVSampleFormat[2];
77 sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
78 sample_fmts[1] = AV_SAMPLE_FMT_NONE;
79 sink_params->sample_fmts = sample_fmts;
81 int64_t* channel_layouts = new int64_t[2];
82 channel_layouts[0] = _channel_layout;
83 channel_layouts[1] = -1;
84 sink_params->channel_layouts = channel_layouts;
86 sink_params->sample_rates = new int[2];
87 sink_params->sample_rates[0] = _sample_rate;
88 sink_params->sample_rates[1] = -1;
94 AudioFilterGraph::src_name () const
100 AudioFilterGraph::sink_name () const
102 return "abuffersink";
106 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
108 int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
109 DCPOMATIC_ASSERT (process_channels >= buffers->channels());
111 if (buffers->channels() < process_channels) {
112 /* We are processing more data than we actually have (see the hack in
113 the constructor) so we need to create new buffers with some extra
116 shared_ptr<AudioBuffers> extended_buffers (new AudioBuffers (process_channels, buffers->frames()));
117 for (int i = 0; i < buffers->channels(); ++i) {
118 extended_buffers->copy_channel_from (buffers.get(), i, i);
120 for (int i = buffers->channels(); i < process_channels; ++i) {
121 extended_buffers->make_silent (i);
124 buffers = extended_buffers;
127 _in_frame->extended_data = new uint8_t*[process_channels];
128 for (int i = 0; i < buffers->channels(); ++i) {
129 if (i < AV_NUM_DATA_POINTERS) {
130 _in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
132 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
135 _in_frame->nb_samples = buffers->frames ();
136 _in_frame->format = AV_SAMPLE_FMT_FLTP;
137 _in_frame->sample_rate = _sample_rate;
138 _in_frame->channel_layout = _channel_layout;
139 _in_frame->channels = process_channels;
141 int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
143 delete[] _in_frame->extended_data;
144 /* Reset extended_data to its original value so that av_frame_free
145 does not try to free it.
147 _in_frame->extended_data = _in_frame->data;
151 av_strerror (r, buffer, sizeof(buffer));
152 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), &buffer[0]));
156 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
160 /* We don't extract audio data here, since the only use of this class
164 av_frame_unref (_frame);