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 #include "dcpomatic_assert.h"
  26 #include "exceptions.h"
  27 extern "C" {
  28 #include <libavfilter/buffersink.h>
  29 #include <libavfilter/buffersrc.h>
  30 #include <libavutil/channel_layout.h>
  31 #include <libavutil/opt.h>
  32 }
  33 #include <iostream>
  34
  35 #include "i18n.h"
  36
  37
  38 using std::cout;
  39 using std::make_shared;
  40 using std::shared_ptr;
  41 using std::string;
  42
  43
  44 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
  45         : _sample_rate (sample_rate)
  46         , _channels (channels)
  47 {
  48         /* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
  49            so we need to tell it we're using 16 channels if we are using more than 8.
  50         */
  51         if (_channels > 8) {
  52                 _channel_layout = av_get_default_channel_layout (16);
  53         } else {
  54                 _channel_layout = av_get_default_channel_layout (_channels);
  55         }
  56
  57         _in_frame = av_frame_alloc ();
  58         if (_in_frame == nullptr) {
  59                 throw std::bad_alloc();
  60         }
  61 }
  62
  63 AudioFilterGraph::~AudioFilterGraph()
  64 {
  65         av_frame_free (&_in_frame);
  66 }
  67
  68 string
  69 AudioFilterGraph::src_parameters () const
  70 {
  71         char layout[64];
  72         av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
  73
  74         char buffer[256];
  75         snprintf (
  76                 buffer, sizeof(buffer), "time_base=1/1:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
  77                 _sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), layout
  78                 );
  79
  80         return buffer;
  81 }
  82
  83
  84 void
  85 AudioFilterGraph::set_parameters (AVFilterContext* context) const
  86 {
  87         AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE };
  88         int r = av_opt_set_int_list (context, "sample_fmts", sample_fmts, AV_SAMPLE_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
  89         DCPOMATIC_ASSERT (r >= 0);
  90
  91         int64_t channel_layouts[] = { _channel_layout, -1 };
  92         r = av_opt_set_int_list (context, "channel_layouts", channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
  93         DCPOMATIC_ASSERT (r >= 0);
  94
  95         int sample_rates[] = { _sample_rate, -1 };
  96         r = av_opt_set_int_list (context, "sample_rates", sample_rates, -1, AV_OPT_SEARCH_CHILDREN);
  97         DCPOMATIC_ASSERT (r >= 0);
  98 }
  99
 100
 101 string
 102 AudioFilterGraph::src_name () const
 103 {
 104         return "abuffer";
 105 }
 106
 107 string
 108 AudioFilterGraph::sink_name () const
 109 {
 110         return "abuffersink";
 111 }
 112
 113 void
 114 AudioFilterGraph::process (shared_ptr<AudioBuffers> buffers)
 115 {
 116         DCPOMATIC_ASSERT (buffers->frames() > 0);
 117         int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
 118         DCPOMATIC_ASSERT (process_channels >= buffers->channels());
 119
 120         if (buffers->channels() < process_channels) {
 121                 /* We are processing more data than we actually have (see the hack in
 122                    the constructor) so we need to create new buffers with some extra
 123                    silent channels.
 124                 */
 125                 auto extended_buffers = make_shared<AudioBuffers>(process_channels, buffers->frames());
 126                 for (int i = 0; i < buffers->channels(); ++i) {
 127                         extended_buffers->copy_channel_from (buffers.get(), i, i);
 128                 }
 129                 for (int i = buffers->channels(); i < process_channels; ++i) {
 130                         extended_buffers->make_silent (i);
 131                 }
 132
 133                 buffers = extended_buffers;
 134         }
 135
 136         _in_frame->extended_data = new uint8_t*[process_channels];
 137         for (int i = 0; i < buffers->channels(); ++i) {
 138                 if (i < AV_NUM_DATA_POINTERS) {
 139                         _in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 140                 }
 141                 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 142         }
 143
 144         _in_frame->nb_samples = buffers->frames ();
 145         _in_frame->format = AV_SAMPLE_FMT_FLTP;
 146         _in_frame->sample_rate = _sample_rate;
 147         _in_frame->channel_layout = _channel_layout;
 148         _in_frame->channels = process_channels;
 149
 150         int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
 151
 152         delete[] _in_frame->extended_data;
 153         /* Reset extended_data to its original value so that av_frame_free
 154            does not try to free it.
 155         */
 156         _in_frame->extended_data = _in_frame->data;
 157
 158         if (r < 0) {
 159                 char buffer[256];
 160                 av_strerror (r, buffer, sizeof(buffer));
 161                 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), &buffer[0]));
 162         }
 163
 164         while (true) {
 165                 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
 166                         break;
 167                 }
 168
 169                 /* We don't extract audio data here, since the only use of this class
 170                    is for ebur128.
 171                 */
 172
 173                 av_frame_unref (_frame);
 174         }
 175 }