git.sesse.net Git - casparcg/blob - core/mixer/audio/audio_mixer.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_mixer.h"\r
  23 \r
  24 #include <core/mixer/write_frame.h>\r
  25 #include <core/producer/frame/audio_transform.h>\r
  26 \r
  27 namespace caspar { namespace core {\r
  28         \r
  29 struct audio_mixer::implementation\r
  30 {\r
  31         std::deque<std::vector<short>> audio_data_;\r
  32         std::stack<core::audio_transform> transform_stack_;\r
  33 \r
  34         std::map<int, core::audio_transform> prev_audio_transforms_;\r
  35         std::map<int, core::audio_transform> next_audio_transforms_;\r
  36 \r
  37 public:\r
  38         implementation()\r
  39         {\r
  40                 transform_stack_.push(core::audio_transform());\r
  41 \r
  42                 // frame delay\r
  43                 audio_data_.push_back(std::vector<short>());\r
  44         }\r
  45         \r
  46         void begin(const core::basic_frame& frame)\r
  47         {\r
  48                 transform_stack_.push(transform_stack_.top()*frame.get_audio_transform());\r
  49         }\r
  50 \r
  51         void visit(const core::write_frame& frame)\r
  52         {\r
  53                 if(!transform_stack_.top().get_has_audio())\r
  54                         return;\r
  55 \r
  56                 auto& audio_data = frame.audio_data();\r
  57                 auto tag = frame.tag(); // Get the identifier for the audio-stream.\r
  58 \r
  59                 if(audio_data_.back().empty())\r
  60                         audio_data_.back().resize(audio_data.size(), 0);\r
  61                 \r
  62                 auto next = transform_stack_.top();\r
  63                 auto prev = next;\r
  64 \r
  65                 auto it = prev_audio_transforms_.find(tag);\r
  66                 if(it != prev_audio_transforms_.end())\r
  67                         prev = it->second;\r
  68                                 \r
  69                 next_audio_transforms_[tag] = next; // Store all active tags, inactive tags will be removed in end_pass.\r
  70                 \r
  71                 \r
  72                 if(next.get_gain() < 0.001 && prev.get_gain() < 0.001)\r
  73                         return;\r
  74                 \r
  75                 static const int BASE = 1<<15;\r
  76 \r
  77                 auto next_gain = static_cast<int>(next.get_gain()*BASE);\r
  78                 auto prev_gain = static_cast<int>(prev.get_gain()*BASE);\r
  79                 \r
  80                 int n_samples = audio_data_.back().size();\r
  81 \r
  82                 tbb::parallel_for\r
  83                 (\r
  84                         tbb::blocked_range<size_t>(0, audio_data.size()),\r
  85                         [&](const tbb::blocked_range<size_t>& r)\r
  86                         {\r
  87                                 for(size_t n = r.begin(); n < r.end(); ++n)\r
  88                                 {\r
  89                                         int sample_gain = (prev_gain - (prev_gain * n)/n_samples) + (next_gain * n)/n_samples;\r
  90                                         \r
  91                                         int sample = (static_cast<int>(audio_data[n])*sample_gain)/BASE;\r
  92                                         \r
  93                                         audio_data_.back()[n] = static_cast<short>((static_cast<int>(audio_data_.back()[n]) + sample) & 0xFFFF);\r
  94                                 }\r
  95                         }\r
  96                 );\r
  97         }\r
  98 \r
  99 \r
 100         void begin(const core::audio_transform& transform)\r
 101         {\r
 102                 transform_stack_.push(transform_stack_.top()*transform);\r
 103         }\r
 104                 \r
 105         void end()\r
 106         {\r
 107                 transform_stack_.pop();\r
 108         }\r
 109 \r
 110         std::vector<short> begin_pass()\r
 111         {\r
 112                 auto result = std::move(audio_data_.front());\r
 113                 audio_data_.pop_front();\r
 114                 \r
 115                 audio_data_.push_back(std::vector<short>());\r
 116 \r
 117                 return result;\r
 118         }\r
 119 \r
 120         void end_pass()\r
 121         {\r
 122                 prev_audio_transforms_ = std::move(next_audio_transforms_);\r
 123         }\r
 124 };\r
 125 \r
 126 audio_mixer::audio_mixer() : impl_(new implementation()){}\r
 127 void audio_mixer::begin(const core::basic_frame& frame){impl_->begin(frame);}\r
 128 void audio_mixer::visit(core::write_frame& frame){impl_->visit(frame);}\r
 129 void audio_mixer::end(){impl_->end();}\r
 130 std::vector<short> audio_mixer::begin_pass(){return impl_->begin_pass();}       \r
 131 void audio_mixer::end_pass(){impl_->end_pass();}\r
 132 \r
 133 }}