87a158a4e356c7af9dd0cd26f19ab6df29668e18
[dcpomatic.git] / src / lib / audio_filter_graph.cc
1 /*
2     Copyright (C) 2015 Carl Hetherington <cth@carlh.net>
3
4     This file is part of DCP-o-matic.
5
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.
10
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.
15
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/>.
18
19 */
20
21 #include "audio_filter_graph.h"
22 #include "audio_buffers.h"
23 #include "compose.hpp"
24 extern "C" {
25 #include <libavfilter/buffersink.h>
26 #include <libavfilter/buffersrc.h>
27 }
28
29 #include "i18n.h"
30
31 using std::string;
32 using std::cout;
33 using boost::shared_ptr;
34
35 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
36         : _sample_rate (sample_rate)
37         , _channels (channels)
38 {
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.
41         */
42         if (_channels > 8) {
43                 _channel_layout = av_get_default_channel_layout (16);
44         } else {
45                 _channel_layout = av_get_default_channel_layout (_channels);
46         }
47
48         _in_frame = av_frame_alloc ();
49 }
50
51 AudioFilterGraph::~AudioFilterGraph()
52 {
53         av_frame_free (&_in_frame);
54 }
55
56 string
57 AudioFilterGraph::src_parameters () const
58 {
59         SafeStringStream a;
60
61         char buffer[64];
62         av_get_channel_layout_string (buffer, sizeof(buffer), 0, _channel_layout);
63
64         a << "time_base=1/1:sample_rate=" << _sample_rate << ":"
65           << "sample_fmt=" << av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP) << ":"
66           << "channel_layout=" << buffer;
67
68         return a.str ();
69 }
70
71 void *
72 AudioFilterGraph::sink_parameters () const
73 {
74         AVABufferSinkParams* sink_params = av_abuffersink_params_alloc ();
75
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;
80
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;
85
86         sink_params->sample_rates = new int[2];
87         sink_params->sample_rates[0] = _sample_rate;
88         sink_params->sample_rates[1] = -1;
89
90         return sink_params;
91 }
92
93 string
94 AudioFilterGraph::src_name () const
95 {
96         return "abuffer";
97 }
98
99 string
100 AudioFilterGraph::sink_name () const
101 {
102         return "abuffersink";
103 }
104
105 void
106 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
107 {
108         int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
109         DCPOMATIC_ASSERT (process_channels >= buffers->channels());
110
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
114                    silent channels.
115                 */
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);
119                 }
120                 for (int i = buffers->channels(); i < process_channels; ++i) {
121                         extended_buffers->make_silent (i);
122                 }
123
124                 buffers = extended_buffers;
125         }
126
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));
131                 }
132                 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
133         }
134
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;
140
141         int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
142
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.
146         */
147         _in_frame->extended_data = _in_frame->data;
148
149         if (r < 0) {
150                 char buffer[256];
151                 av_strerror (r, buffer, sizeof(buffer));
152                 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), buffer));
153         }
154
155         while (true) {
156                 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
157                         break;
158                 }
159
160                 /* We don't extract audio data here, since the only use of this class
161                    is for ebur128.
162                 */
163
164                 av_frame_unref (_frame);
165         }
166 }