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
  22 #include "audio_buffers.h"
  23 #include "audio_filter_graph.h"
  24 #include "compose.hpp"
  25 extern "C" {
  26 #include <libavfilter/buffersink.h>
  27 #include <libavfilter/buffersrc.h>
  28 #include <libavutil/channel_layout.h>
  29 #include <libavutil/opt.h>
  30 }
  31 #include <iostream>
  32
  33 #include "i18n.h"
  34
  35
  36 using std::cout;
  37 using std::make_shared;
  38 using std::shared_ptr;
  39 using std::string;
  40
  41
  42 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
  43         : _sample_rate (sample_rate)
  44         , _channels (channels)
  45 {
  46         /* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
  47            so we need to tell it we're using 16 channels if we are using more than 8.
  48         */
  49         if (_channels > 8) {
  50                 _channel_layout = av_get_default_channel_layout (16);
  51         } else {
  52                 _channel_layout = av_get_default_channel_layout (_channels);
  53         }
  54
  55         _in_frame = av_frame_alloc ();
  56 }
  57
  58 AudioFilterGraph::~AudioFilterGraph()
  59 {
  60         av_frame_free (&_in_frame);
  61 }
  62
  63 string
  64 AudioFilterGraph::src_parameters () const
  65 {
  66         char layout[64];
  67         av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
  68
  69         char buffer[256];
  70         snprintf (
  71                 buffer, sizeof(buffer), "time_base=1/1:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
  72                 _sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), layout
  73                 );
  74
  75         return buffer;
  76 }
  77
  78
  79 void
  80 AudioFilterGraph::set_parameters (AVFilterContext* context) const
  81 {
  82         AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
  83         int r = av_opt_set_int_list (context, "sample_fmts", sample_fmts, AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
  84         DCPOMATIC_ASSERT (r >= 0);
  85
  86         int64_t channel_layouts[] = { _channel_layout, -1 };
  87         r = av_opt_set_int_list (context, "channel_layouts", channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
  88         DCPOMATIC_ASSERT (r >= 0);
  89
  90         int sample_rates[] = { _sample_rate, -1 };
  91         r = av_opt_set_int_list (context, "sample_rates", sample_rates, -1, AV_OPT_SEARCH_CHILDREN);
  92         DCPOMATIC_ASSERT (r >= 0);
  93 }
  94
  95
  96 string
  97 AudioFilterGraph::src_name () const
  98 {
  99         return "abuffer";
 100 }
 101
 102 string
 103 AudioFilterGraph::sink_name () const
 104 {
 105         return "abuffersink";
 106 }
 107
 108 void
 109 AudioFilterGraph::process (shared_ptr<AudioBuffers> buffers)
 110 {
 111         DCPOMATIC_ASSERT (buffers->frames() > 0);
 112         int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
 113         DCPOMATIC_ASSERT (process_channels >= buffers->channels());
 114
 115         if (buffers->channels() < process_channels) {
 116                 /* We are processing more data than we actually have (see the hack in
 117                    the constructor) so we need to create new buffers with some extra
 118                    silent channels.
 119                 */
 120                 auto extended_buffers = make_shared<AudioBuffers>(process_channels, buffers->frames());
 121                 for (int i = 0; i < buffers->channels(); ++i) {
 122                         extended_buffers->copy_channel_from (buffers.get(), i, i);
 123                 }
 124                 for (int i = buffers->channels(); i < process_channels; ++i) {
 125                         extended_buffers->make_silent (i);
 126                 }
 127
 128                 buffers = extended_buffers;
 129         }
 130
 131         _in_frame->extended_data = new uint8_t*[process_channels];
 132         for (int i = 0; i < buffers->channels(); ++i) {
 133                 if (i < AV_NUM_DATA_POINTERS) {
 134                         _in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 135                 }
 136                 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 137         }
 138
 139         _in_frame->nb_samples = buffers->frames ();
 140         _in_frame->format = AV_SAMPLE_FMT_FLTP;
 141         _in_frame->sample_rate = _sample_rate;
 142         _in_frame->channel_layout = _channel_layout;
 143         _in_frame->channels = process_channels;
 144
 145         int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
 146
 147         delete[] _in_frame->extended_data;
 148         /* Reset extended_data to its original value so that av_frame_free
 149            does not try to free it.
 150         */
 151         _in_frame->extended_data = _in_frame->data;
 152
 153         if (r < 0) {
 154                 char buffer[256];
 155                 av_strerror (r, buffer, sizeof(buffer));
 156                 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), &buffer[0]));
 157         }
 158
 159         while (true) {
 160                 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
 161                         break;
 162                 }
 163
 164                 /* We don't extract audio data here, since the only use of this class
 165                    is for ebur128.
 166                 */
 167
 168                 av_frame_unref (_frame);
 169         }
 170 }