2 Copyright (C) 2015 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_filter_graph.h"
21 #include "audio_buffers.h"
22 #include "compose.hpp"
24 #include <libavfilter/buffersink.h>
25 #include <libavfilter/buffersrc.h>
32 using boost::shared_ptr;
34 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
35 : _sample_rate (sample_rate)
36 , _channels (channels)
38 /* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
39 so we need to tell it we're using 16 channels if we are using more than 8.
42 _channel_layout = av_get_default_channel_layout (16);
44 _channel_layout = av_get_default_channel_layout (_channels);
47 _in_frame = av_frame_alloc ();
50 AudioFilterGraph::~AudioFilterGraph()
52 av_frame_free (&_in_frame);
56 AudioFilterGraph::src_parameters () const
61 av_get_channel_layout_string (buffer, sizeof(buffer), 0, _channel_layout);
63 a << "time_base=1/1:sample_rate=" << _sample_rate << ":"
64 << "sample_fmt=" << av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP) << ":"
65 << "channel_layout=" << buffer;
71 AudioFilterGraph::sink_parameters () const
73 AVABufferSinkParams* sink_params = av_abuffersink_params_alloc ();
75 AVSampleFormat* sample_fmts = new AVSampleFormat[2];
76 sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
77 sample_fmts[1] = AV_SAMPLE_FMT_NONE;
78 sink_params->sample_fmts = sample_fmts;
80 int64_t* channel_layouts = new int64_t[2];
81 channel_layouts[0] = _channel_layout;
82 channel_layouts[1] = -1;
83 sink_params->channel_layouts = channel_layouts;
85 sink_params->sample_rates = new int[2];
86 sink_params->sample_rates[0] = _sample_rate;
87 sink_params->sample_rates[1] = -1;
93 AudioFilterGraph::src_name () const
99 AudioFilterGraph::sink_name () const
101 return "abuffersink";
105 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
107 int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
108 DCPOMATIC_ASSERT (process_channels >= buffers->channels());
110 if (buffers->channels() < process_channels) {
111 /* We are processing more data than we actually have (see the hack in
112 the constructor) so we need to create new buffers with some extra
115 shared_ptr<AudioBuffers> extended_buffers (new AudioBuffers (process_channels, buffers->frames()));
116 for (int i = 0; i < buffers->channels(); ++i) {
117 extended_buffers->copy_channel_from (buffers.get(), i, i);
119 for (int i = buffers->channels(); i < process_channels; ++i) {
120 extended_buffers->make_silent (i);
123 buffers = extended_buffers;
126 _in_frame->extended_data = new uint8_t*[process_channels];
127 for (int i = 0; i < buffers->channels(); ++i) {
128 if (i < AV_NUM_DATA_POINTERS) {
129 _in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
131 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
134 _in_frame->nb_samples = buffers->frames ();
135 _in_frame->format = AV_SAMPLE_FMT_FLTP;
136 _in_frame->sample_rate = _sample_rate;
137 _in_frame->channel_layout = _channel_layout;
138 _in_frame->channels = process_channels;
140 int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
142 delete[] _in_frame->extended_data;
143 /* Reset extended_data to its original value so that av_frame_free
144 does not try to free it.
146 _in_frame->extended_data = _in_frame->data;
150 av_strerror (r, buffer, sizeof(buffer));
151 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), buffer));
155 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
159 /* We don't extract audio data here, since the only use of this class
163 av_frame_unref (_frame);