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 #include <libavutil/channel_layout.h>
  28 }
  29 #include <iostream>
  30
  31 #include "i18n.h"
  32
  33 using std::string;
  34 using std::cout;
  35 using boost::shared_ptr;
  36
  37 AudioFilterGraph::AudioFilterGraph (int sample_rate, int channels)
  38         : _sample_rate (sample_rate)
  39         , _channels (channels)
  40 {
  41         /* FFmpeg doesn't know any channel layouts for any counts between 8 and 16,
  42            so we need to tell it we're using 16 channels if we are using more than 8.
  43         */
  44         if (_channels > 8) {
  45                 _channel_layout = av_get_default_channel_layout (16);
  46         } else {
  47                 _channel_layout = av_get_default_channel_layout (_channels);
  48         }
  49
  50         _in_frame = av_frame_alloc ();
  51 }
  52
  53 AudioFilterGraph::~AudioFilterGraph()
  54 {
  55         av_frame_free (&_in_frame);
  56 }
  57
  58 string
  59 AudioFilterGraph::src_parameters () const
  60 {
  61         char layout[64];
  62         av_get_channel_layout_string (layout, sizeof(layout), 0, _channel_layout);
  63
  64         char buffer[256];
  65         snprintf (
  66                 buffer, sizeof(buffer), "time_base=1/1:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
  67                 _sample_rate, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), layout
  68                 );
  69
  70         return buffer;
  71 }
  72
  73 void *
  74 AudioFilterGraph::sink_parameters () const
  75 {
  76         AVABufferSinkParams* sink_params = av_abuffersink_params_alloc ();
  77
  78         AVSampleFormat* sample_fmts = new AVSampleFormat[2];
  79         sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
  80         sample_fmts[1] = AV_SAMPLE_FMT_NONE;
  81         sink_params->sample_fmts = sample_fmts;
  82
  83         int64_t* channel_layouts = new int64_t[2];
  84         channel_layouts[0] = _channel_layout;
  85         channel_layouts[1] = -1;
  86         sink_params->channel_layouts = channel_layouts;
  87
  88         sink_params->sample_rates = new int[2];
  89         sink_params->sample_rates[0] = _sample_rate;
  90         sink_params->sample_rates[1] = -1;
  91
  92         return sink_params;
  93 }
  94
  95 string
  96 AudioFilterGraph::src_name () const
  97 {
  98         return "abuffer";
  99 }
 100
 101 string
 102 AudioFilterGraph::sink_name () const
 103 {
 104         return "abuffersink";
 105 }
 106
 107 void
 108 AudioFilterGraph::process (shared_ptr<const AudioBuffers> buffers)
 109 {
 110         DCPOMATIC_ASSERT (buffers->frames() > 0);
 111         int const process_channels = av_get_channel_layout_nb_channels (_channel_layout);
 112         DCPOMATIC_ASSERT (process_channels >= buffers->channels());
 113
 114         if (buffers->channels() < process_channels) {
 115                 /* We are processing more data than we actually have (see the hack in
 116                    the constructor) so we need to create new buffers with some extra
 117                    silent channels.
 118                 */
 119                 shared_ptr<AudioBuffers> extended_buffers (new AudioBuffers (process_channels, buffers->frames()));
 120                 for (int i = 0; i < buffers->channels(); ++i) {
 121                         extended_buffers->copy_channel_from (buffers.get(), i, i);
 122                 }
 123                 for (int i = buffers->channels(); i < process_channels; ++i) {
 124                         extended_buffers->make_silent (i);
 125                 }
 126
 127                 buffers = extended_buffers;
 128         }
 129
 130         _in_frame->extended_data = new uint8_t*[process_channels];
 131         for (int i = 0; i < buffers->channels(); ++i) {
 132                 if (i < AV_NUM_DATA_POINTERS) {
 133                         _in_frame->data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 134                 }
 135                 _in_frame->extended_data[i] = reinterpret_cast<uint8_t*> (buffers->data(i));
 136         }
 137
 138         _in_frame->nb_samples = buffers->frames ();
 139         _in_frame->format = AV_SAMPLE_FMT_FLTP;
 140         _in_frame->sample_rate = _sample_rate;
 141         _in_frame->channel_layout = _channel_layout;
 142         _in_frame->channels = process_channels;
 143
 144         int r = av_buffersrc_write_frame (_buffer_src_context, _in_frame);
 145
 146         delete[] _in_frame->extended_data;
 147         /* Reset extended_data to its original value so that av_frame_free
 148            does not try to free it.
 149         */
 150         _in_frame->extended_data = _in_frame->data;
 151
 152         if (r < 0) {
 153                 char buffer[256];
 154                 av_strerror (r, buffer, sizeof(buffer));
 155                 throw DecodeError (String::compose (N_("could not push buffer into filter chain (%1)"), &buffer[0]));
 156         }
 157
 158         while (true) {
 159                 if (av_buffersink_get_frame (_buffer_sink_context, _frame) < 0) {
 160                         break;
 161                 }
 162
 163                 /* We don't extract audio data here, since the only use of this class
 164                    is for ebur128.
 165                 */
 166
 167                 av_frame_unref (_frame);
 168         }
 169 }