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