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

   1 #include "../../../stdafx.h"\r
   2 \r
   3 #include "audio_decoder.h"\r
   4 \r
   5 #include "../../../../common/utility/memory.h"\r
   6 \r
   7 #include <queue>\r
   8 \r
   9 #include <tbb/cache_aligned_allocator.h>\r
  10                 \r
  11 #if defined(_MSC_VER)\r
  12 #pragma warning (push)\r
  13 #pragma warning (disable : 4244)\r
  14 #endif\r
  15 extern "C" \r
  16 {\r
  17         #define __STDC_CONSTANT_MACROS\r
  18         #define __STDC_LIMIT_MACROS\r
  19         #include <libavformat/avformat.h>\r
  20         #include <libavcodec/avcodec.h>\r
  21 }\r
  22 #if defined(_MSC_VER)\r
  23 #pragma warning (pop)\r
  24 #endif\r
  25 \r
  26 namespace caspar { namespace core { namespace ffmpeg{\r
  27 \r
  28 struct audio_decoder::implementation : boost::noncopyable\r
  29 {\r
  30         static const int FRAME_AUDIO_SAMPLES = 1920*2;\r
  31         static const int SAMPLE_RATE = 48000;\r
  32 \r
  33         implementation(AVCodecContext* codec_context) \r
  34                 : current_chunk_(), codec_context_(codec_context), audio_resample_buffer_(4*SAMPLE_RATE*2+FF_INPUT_BUFFER_PADDING_SIZE/2),\r
  35                 audio_buffer_(4*SAMPLE_RATE*2+FF_INPUT_BUFFER_PADDING_SIZE/2)/*, resample_context_(nullptr)*/\r
  36         {\r
  37                 //if(codec_context_->sample_rate != SAMPLE_RATE)\r
  38                 //{\r
  39                 //      resample_context_ = av_audio_resample_init \r
  40                 //      (\r
  41                 //              2, \r
  42                 //              codec_context_->channels,\r
  43                 //              SAMPLE_RATE,    // out rate\r
  44                 //              codec_context_->sample_rate,    // in rate\r
  45                 //              SAMPLE_FMT_S16,\r
  46                 //              codec_context_->sample_fmt,\r
  47                 //              16, 10, 0, 1.0\r
  48                 //      );      \r
  49                 //}\r
  50         }\r
  51 \r
  52         ~implementation()\r
  53         {\r
  54                 //if(resample_context_ != nullptr)\r
  55                 //      audio_resample_close(resample_context_);\r
  56         }\r
  57                 \r
  58         std::vector<std::vector<short>> execute(const aligned_buffer& audio_packet)\r
  59         {                       \r
  60                 static const std::vector<std::vector<short>> silence(1920*2, 0);\r
  61                 \r
  62                 int written_bytes = audio_buffer_.size()*2 - FF_INPUT_BUFFER_PADDING_SIZE;\r
  63                 const int result = avcodec_decode_audio2(codec_context_, audio_buffer_.data(), &written_bytes, audio_packet.data(), audio_packet.size());\r
  64 \r
  65                 if(result <= 0)\r
  66                         return silence;\r
  67                                 \r
  68                 if(codec_context_->sample_rate != SAMPLE_RATE)\r
  69                 {\r
  70                         //if(resample_context_ == nullptr)\r
  71                                 return silence;\r
  72                         \r
  73                         //int samples_output = audio_resample\r
  74                         //      ( \r
  75                         //              resample_context_,\r
  76                         //              audio_resample_buffer_.data(),\r
  77                         //              audio_buffer_.data(),\r
  78                         //              written_bytes/2 // in samples\r
  79                         //      );\r
  80 \r
  81                         //if(samples_output == -1)\r
  82                         //{     \r
  83                         //      CASPAR_LOG(trace) << "Resampling error";\r
  84                         //      audio_resample_close(resample_context_);\r
  85                         //      resample_context_ = nullptr;\r
  86                         //}\r
  87 \r
  88                         //current_chunk_.insert(current_chunk_.end(), audio_resample_buffer_.data(), audio_resample_buffer_.data() + samples_output);\r
  89                 }\r
  90                 else\r
  91                         current_chunk_.insert(current_chunk_.end(), audio_buffer_.data(), audio_buffer_.data() + written_bytes/2);\r
  92 \r
  93                 std::vector<std::vector<short>> chunks_;\r
  94                                 \r
  95                 const auto last = current_chunk_.end() - current_chunk_.size() % FRAME_AUDIO_SAMPLES;\r
  96 \r
  97                 for(auto it = current_chunk_.begin(); it != last; it += FRAME_AUDIO_SAMPLES)            \r
  98                         chunks_.push_back(std::vector<short>(it, it + FRAME_AUDIO_SAMPLES));            \r
  99 \r
 100                 current_chunk_.erase(current_chunk_.begin(), last);\r
 101                 \r
 102                 return chunks_;\r
 103         }\r
 104 \r
 105         //ReSampleContext* resample_context_;\r
 106                                                 \r
 107         typedef std::vector<short, tbb::cache_aligned_allocator<short>> buffer;\r
 108 \r
 109         buffer audio_buffer_;   \r
 110         buffer audio_resample_buffer_;\r
 111 \r
 112         std::deque<short, tbb::cache_aligned_allocator<short>> current_chunk_;\r
 113 \r
 114         std::vector<short, tbb::cache_aligned_allocator<short>> current_resample_chunk_;\r
 115 \r
 116         AVCodecContext* codec_context_;\r
 117 };\r
 118 \r
 119 audio_decoder::audio_decoder(AVCodecContext* codec_context) : impl_(new implementation(codec_context)){}\r
 120 std::vector<std::vector<short>> audio_decoder::execute(const aligned_buffer& audio_packet){return impl_->execute(audio_packet);}\r
 121 }}}