git.sesse.net Git - casparcg/blob - modules/ffmpeg/producer/audio/audio_decoder.cpp

   1 /*\r
   2 * copyright (c) 2010 Sveriges Television AB <info@casparcg.com>\r
   3 *\r
   4 *  This file is part of CasparCG.\r
   5 *\r
   6 *    CasparCG is free software: you can redistribute it and/or modify\r
   7 *    it under the terms of the GNU General Public License as published by\r
   8 *    the Free Software Foundation, either version 3 of the License, or\r
   9 *    (at your option) any later version.\r
  10 *\r
  11 *    CasparCG is distributed in the hope that it will be useful,\r
  12 *    but WITHOUT ANY WARRANTY; without even the implied warranty of\r
  13 *    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the\r
  14 *    GNU General Public License for more details.\r
  15 \r
  16 *    You should have received a copy of the GNU General Public License\r
  17 *    along with CasparCG.  If not, see <http://www.gnu.org/licenses/>.\r
  18 *\r
  19 */\r
  20 #include "../../stdafx.h"\r
  21 \r
  22 #include "audio_decoder.h"\r
  23 \r
  24 #include "../../ffmpeg_error.h"\r
  25 \r
  26 #include <tbb/task_group.h>\r
  27 \r
  28 #if defined(_MSC_VER)\r
  29 #pragma warning (push)\r
  30 #pragma warning (disable : 4244)\r
  31 #endif\r
  32 extern "C" \r
  33 {\r
  34         #define __STDC_CONSTANT_MACROS\r
  35         #define __STDC_LIMIT_MACROS\r
  36         #include <libavformat/avformat.h>\r
  37         #include <libavcodec/avcodec.h>\r
  38 }\r
  39 #if defined(_MSC_VER)\r
  40 #pragma warning (pop)\r
  41 #endif\r
  42 \r
  43 namespace caspar {\r
  44         \r
  45 struct audio_decoder::implementation : boost::noncopyable\r
  46 {       \r
  47         std::shared_ptr<AVCodecContext>                                                         codec_context_;         \r
  48         const core::video_format_desc                                                           format_desc_;\r
  49         int                                                                                                                     index_;\r
  50         std::shared_ptr<ReSampleContext>                                                        resampler_;\r
  51 \r
  52         std::vector<int8_t,  tbb::cache_aligned_allocator<int8_t>>      buffer1_;\r
  53         std::vector<int8_t,  tbb::cache_aligned_allocator<int8_t>>      buffer2_;\r
  54         std::vector<int16_t, tbb::cache_aligned_allocator<int16_t>>     audio_samples_; \r
  55         std::queue<std::shared_ptr<AVPacket>>                                           packets_;\r
  56 \r
  57         int64_t                                                                                                         nb_frames_;\r
  58 public:\r
  59         explicit implementation(const std::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) \r
  60                 : format_desc_(format_desc)     \r
  61                 , nb_frames_(0)\r
  62                 , buffer1_(AVCODEC_MAX_AUDIO_FRAME_SIZE*2)\r
  63         {                                       \r
  64                 try\r
  65                 {\r
  66                         AVCodec* dec;\r
  67                         index_ = THROW_ON_ERROR2(av_find_best_stream(context.get(), AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0), "[audio_decoder]");\r
  68 \r
  69                         THROW_ON_ERROR2(avcodec_open(context->streams[index_]->codec, dec), "[audio_decoder]");\r
  70                 }\r
  71                 catch(...)\r
  72                 {\r
  73                         CASPAR_LOG_CURRENT_EXCEPTION();\r
  74                         CASPAR_LOG(warning) << "[audio_decoder] Failed to open audio. Running without audio.";\r
  75                         return;\r
  76                 }\r
  77 \r
  78                 codec_context_.reset(context->streams[index_]->codec, avcodec_close);\r
  79 \r
  80                 //nb_frames_ = context->streams[index_]->nb_frames;\r
  81                 //if(nb_frames_ == 0)\r
  82                 //      nb_frames_ = context->streams[index_]->duration * context->streams[index_]->time_base.den;\r
  83 \r
  84                 if(codec_context_ &&\r
  85                    (codec_context_->sample_rate != static_cast<int>(format_desc_.audio_sample_rate) || \r
  86                     codec_context_->channels    != static_cast<int>(format_desc_.audio_channels)) ||\r
  87                         codec_context_->sample_fmt      != AV_SAMPLE_FMT_S16)\r
  88                 {       \r
  89                         auto resampler = av_audio_resample_init(format_desc_.audio_channels,    codec_context_->channels,\r
  90                                                                                                         format_desc_.audio_sample_rate, codec_context_->sample_rate,\r
  91                                                                                                         AV_SAMPLE_FMT_S16,                              codec_context_->sample_fmt,\r
  92                                                                                                         16, 10, 0, 0.8);\r
  93 \r
  94                         buffer2_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\r
  95 \r
  96                         CASPAR_LOG(warning) << L" Invalid audio format. Resampling.";\r
  97 \r
  98                         if(resampler)\r
  99                                 resampler_.reset(resampler, audio_resample_close);\r
 100                         else\r
 101                                 codec_context_ = nullptr;\r
 102                 }               \r
 103         }\r
 104 \r
 105         void push(const std::shared_ptr<AVPacket>& packet)\r
 106         {                       \r
 107                 if(!codec_context_)\r
 108                         return;\r
 109 \r
 110                 if(packet && packet->stream_index != index_)\r
 111                         return;\r
 112 \r
 113                 packets_.push(packet);\r
 114         }       \r
 115         \r
 116         std::vector<std::shared_ptr<std::vector<int16_t>>> poll()\r
 117         {\r
 118                 std::vector<std::shared_ptr<std::vector<int16_t>>> result;\r
 119 \r
 120                 if(!codec_context_)\r
 121                         result.push_back(std::make_shared<std::vector<int16_t>>(format_desc_.audio_samples_per_frame, 0));\r
 122                 else if(!packets_.empty())\r
 123                 {               \r
 124                         auto packet = packets_.front();\r
 125 \r
 126                         if(packet)              \r
 127                         {\r
 128                                 result.push_back(decode(*packet));\r
 129                                 if(packet->size == 0)                                   \r
 130                                         packets_.pop();\r
 131                         }\r
 132                         else                    \r
 133                         {       \r
 134                                 avcodec_flush_buffers(codec_context_.get());\r
 135                                 result.push_back(nullptr);\r
 136                                 packets_.pop();\r
 137                         }\r
 138                 }\r
 139 \r
 140                 return result;\r
 141         }\r
 142 \r
 143         std::shared_ptr<std::vector<int16_t>> decode(AVPacket& pkt)\r
 144         {               \r
 145                 int written_bytes = buffer1_.size() - FF_INPUT_BUFFER_PADDING_SIZE;\r
 146 \r
 147                 int ret = THROW_ON_ERROR2(avcodec_decode_audio3(codec_context_.get(), reinterpret_cast<int16_t*>(buffer1_.data()), &written_bytes, &pkt), "[audio_decoder]");\r
 148 \r
 149                 // There might be several frames in one packet.\r
 150                 pkt.size -= ret;\r
 151                 pkt.data += ret;\r
 152                         \r
 153                 if(resampler_)\r
 154                 {\r
 155                         auto ret = audio_resample(resampler_.get(),\r
 156                                                                                 reinterpret_cast<short*>(buffer2_.data()), \r
 157                                                                                 reinterpret_cast<short*>(buffer1_.data()), \r
 158                                                                                 written_bytes / (av_get_bytes_per_sample(codec_context_->sample_fmt) * codec_context_->channels)); \r
 159                         written_bytes = ret * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * format_desc_.audio_channels;\r
 160                         std::swap(buffer1_, buffer2_);\r
 161                 }\r
 162 \r
 163                 const auto n_samples = written_bytes / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);\r
 164                 const auto samples = reinterpret_cast<int16_t*>(buffer1_.data());\r
 165 \r
 166                 return std::make_shared<std::vector<int16_t>>(samples, samples + n_samples);\r
 167         }\r
 168 \r
 169         bool ready() const\r
 170         {\r
 171                 return !codec_context_ || !packets_.empty();\r
 172         }\r
 173 };\r
 174 \r
 175 audio_decoder::audio_decoder(const std::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) : impl_(new implementation(context, format_desc)){}\r
 176 void audio_decoder::push(const std::shared_ptr<AVPacket>& packet){impl_->push(packet);}\r
 177 bool audio_decoder::ready() const{return impl_->ready();}\r
 178 std::vector<std::shared_ptr<std::vector<int16_t>>> audio_decoder::poll(){return impl_->poll();}\r
 179 int64_t audio_decoder::nb_frames() const{return impl_->nb_frames_;}\r
 180 }