]> git.sesse.net Git - casparcg/blobdiff - modules/ffmpeg/producer/audio/audio_decoder.cpp
[ffmpeg] Fixed compilation problems in Linux
[casparcg] / modules / ffmpeg / producer / audio / audio_decoder.cpp
index 3cd99a38f685aac411bf01c348a98c484f15b1f1..fe69a59cf6aa459f8667ebaf2bdf0864d99bb00e 100644 (file)
-/*\r
-* copyright (c) 2010 Sveriges Television AB <info@casparcg.com>\r
-*\r
-*  This file is part of CasparCG.\r
-*\r
-*    CasparCG is free software: you can redistribute it and/or modify\r
-*    it under the terms of the GNU General Public License as published by\r
-*    the Free Software Foundation, either version 3 of the License, or\r
-*    (at your option) any later version.\r
-*\r
-*    CasparCG is distributed in the hope that it will be useful,\r
-*    but WITHOUT ANY WARRANTY; without even the implied warranty of\r
-*    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the\r
-*    GNU General Public License for more details.\r
-\r
-*    You should have received a copy of the GNU General Public License\r
-*    along with CasparCG.  If not, see <http://www.gnu.org/licenses/>.\r
-*\r
-*/\r
-#include "../../stdafx.h"\r
-\r
-#include "audio_decoder.h"\r
-\r
-#include <tbb/task_group.h>\r
-\r
-#if defined(_MSC_VER)\r
-#pragma warning (push)\r
-#pragma warning (disable : 4244)\r
-#endif\r
-extern "C" \r
-{\r
-       #define __STDC_CONSTANT_MACROS\r
-       #define __STDC_LIMIT_MACROS\r
-       #include <libavformat/avformat.h>\r
-       #include <libavcodec/avcodec.h>\r
-}\r
-#if defined(_MSC_VER)\r
-#pragma warning (pop)\r
-#endif\r
-\r
-namespace caspar {\r
-       \r
-struct audio_decoder::implementation : boost::noncopyable\r
-{      \r
-       std::shared_ptr<AVCodecContext>                                                         codec_context_;         \r
-       const core::video_format_desc                                                           format_desc_;\r
-       int                                                                                                                     index_;\r
-       std::shared_ptr<ReSampleContext>                                                        resampler_;\r
-\r
-       std::vector<int8_t,  tbb::cache_aligned_allocator<int8_t>>      buffer1_;\r
-       std::vector<int8_t,  tbb::cache_aligned_allocator<int8_t>>      buffer2_;\r
-       std::vector<int16_t, tbb::cache_aligned_allocator<int16_t>>     audio_samples_; \r
-       std::queue<std::shared_ptr<AVPacket>>                                           packets_;\r
-\r
-       int64_t                                                                                                         nb_frames_;\r
-public:\r
-       explicit implementation(const std::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) \r
-               : format_desc_(format_desc)     \r
-               , nb_frames_(0)\r
-       {                          \r
-               AVCodec* dec;\r
-               index_ = av_find_best_stream(context.get(), AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);\r
-\r
-               if(index_ < 0)\r
-                       return;\r
-\r
-               int errn = avcodec_open(context->streams[index_]->codec, dec);\r
-               if(errn < 0)\r
-                       return;\r
-                               \r
-               codec_context_.reset(context->streams[index_]->codec, avcodec_close);\r
-\r
-               //nb_frames_ = context->streams[index_]->nb_frames;\r
-               //if(nb_frames_ == 0)\r
-               //      nb_frames_ = context->streams[index_]->duration * context->streams[index_]->time_base.den;\r
-\r
-               if(codec_context_ &&\r
-                  (codec_context_->sample_rate != static_cast<int>(format_desc_.audio_sample_rate) || \r
-                   codec_context_->channels    != static_cast<int>(format_desc_.audio_channels)) ||\r
-                       codec_context_->sample_fmt      != AV_SAMPLE_FMT_S16)\r
-               {       \r
-                       auto resampler = av_audio_resample_init(format_desc_.audio_channels,    codec_context_->channels,\r
-                                                                                                       format_desc_.audio_sample_rate, codec_context_->sample_rate,\r
-                                                                                                       AV_SAMPLE_FMT_S16,                              codec_context_->sample_fmt,\r
-                                                                                                       16, 10, 0, 0.8);\r
-\r
-                       CASPAR_LOG(warning) << L" Invalid audio format. Resampling.";\r
-\r
-                       if(resampler)\r
-                               resampler_.reset(resampler, audio_resample_close);\r
-                       else\r
-                               codec_context_ = nullptr;\r
-               }               \r
-       }\r
-\r
-       void push(const std::shared_ptr<AVPacket>& packet)\r
-       {                       \r
-               if(!codec_context_)\r
-                       return;\r
-\r
-               if(packet && packet->stream_index != index_)\r
-                       return;\r
-\r
-               packets_.push(packet);\r
-       }       \r
-       \r
-       std::vector<std::shared_ptr<std::vector<int16_t>>> poll()\r
-       {\r
-               std::vector<std::shared_ptr<std::vector<int16_t>>> result;\r
-\r
-               if(!codec_context_)\r
-                       result.push_back(std::make_shared<std::vector<int16_t>>(format_desc_.audio_samples_per_frame, 0));\r
-               else if(!packets_.empty())\r
-               {               \r
-                       auto packet = std::move(packets_.front());\r
-                       packets_.pop();\r
-\r
-                       if(packet)              \r
-                       {\r
-                               AVPacket pkt;\r
-                               av_init_packet(&pkt);\r
-                               pkt.data = packet->data;\r
-                               pkt.size = packet->size;\r
-\r
-                               for(int n = 0; n < 64 && pkt.size > 0; ++n)\r
-                                       result.push_back(decode(pkt));\r
-                       }\r
-                       else                    \r
-                       {       \r
-                               avcodec_flush_buffers(codec_context_.get());\r
-                               result.push_back(nullptr);\r
-                       }\r
-               }\r
-\r
-               return result;\r
-       }\r
-\r
-       std::shared_ptr<std::vector<int16_t>> decode(AVPacket& pkt)\r
-       {               \r
-               buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2, 0);\r
-               int written_bytes = buffer1_.size() - FF_INPUT_BUFFER_PADDING_SIZE;\r
-\r
-               const int ret = avcodec_decode_audio3(codec_context_.get(), reinterpret_cast<int16_t*>(buffer1_.data()), &written_bytes, &pkt);\r
-               if(ret < 0)\r
-               {       \r
-                       BOOST_THROW_EXCEPTION(\r
-                               invalid_operation() <<\r
-                               boost::errinfo_api_function("avcodec_decode_audio2") <<\r
-                               boost::errinfo_errno(AVUNERROR(ret)));\r
-               }\r
-\r
-               // There might be several frames in one packet.\r
-               pkt.size -= ret;\r
-               pkt.data += ret;\r
-                       \r
-               buffer1_.resize(written_bytes);\r
-\r
-               if(resampler_)\r
-               {\r
-                       buffer2_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2, 0);\r
-                       auto ret = audio_resample(resampler_.get(),\r
-                                                                               reinterpret_cast<short*>(buffer2_.data()), \r
-                                                                               reinterpret_cast<short*>(buffer1_.data()), \r
-                                                                               buffer1_.size() / (av_get_bytes_per_sample(codec_context_->sample_fmt) * codec_context_->channels)); \r
-                       buffer2_.resize(ret * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * format_desc_.audio_channels);\r
-                       std::swap(buffer1_, buffer2_);\r
-               }\r
-\r
-               const auto n_samples = buffer1_.size() / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);\r
-               const auto samples = reinterpret_cast<int16_t*>(buffer1_.data());\r
-\r
-               return std::make_shared<std::vector<int16_t>>(samples, samples + n_samples);\r
-       }\r
-\r
-       bool ready() const\r
-       {\r
-               return !codec_context_ || !packets_.empty();\r
-       }\r
-};\r
-\r
-audio_decoder::audio_decoder(const std::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) : impl_(new implementation(context, format_desc)){}\r
-void audio_decoder::push(const std::shared_ptr<AVPacket>& packet){impl_->push(packet);}\r
-bool audio_decoder::ready() const{return impl_->ready();}\r
-std::vector<std::shared_ptr<std::vector<int16_t>>> audio_decoder::poll(){return impl_->poll();}\r
-int64_t audio_decoder::nb_frames() const{return impl_->nb_frames_;}\r
-}
\ No newline at end of file
+/*
+* Copyright 2013 Sveriges Television AB http://casparcg.com/
+*
+* This file is part of CasparCG (www.casparcg.com).
+*
+* CasparCG is free software: you can redistribute it and/or modify
+* it under the terms of the GNU General Public License as published by
+* the Free Software Foundation, either version 3 of the License, or
+* (at your option) any later version.
+*
+* CasparCG is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+* GNU General Public License for more details.
+*
+* You should have received a copy of the GNU General Public License
+* along with CasparCG. If not, see <http://www.gnu.org/licenses/>.
+*
+* Author: Robert Nagy, ronag89@gmail.com
+*/
+
+#include "../../StdAfx.h"
+
+#include "audio_decoder.h"
+
+#include "../util/util.h"
+#include "../../ffmpeg_error.h"
+
+#include <core/video_format.h>
+#include <core/mixer/audio/audio_util.h>
+
+#include <common/cache_aligned_vector.h>
+
+#include <queue>
+
+#if defined(_MSC_VER)
+#pragma warning (push)
+#pragma warning (disable : 4244)
+#endif
+extern "C" 
+{
+       #include <libavformat/avformat.h>
+       #include <libavcodec/avcodec.h>
+       #include <libswresample/swresample.h>
+}
+#if defined(_MSC_VER)
+#pragma warning (pop)
+#endif
+
+namespace caspar { namespace ffmpeg {
+       
+struct audio_decoder::implementation : boost::noncopyable
+{      
+       int                                                                             index_                          = -1;
+       const spl::shared_ptr<AVCodecContext>   codec_context_;
+       const int                                                               out_samplerate_;
+       
+       cache_aligned_vector<int32_t>                   buffer_;
+
+       std::queue<spl::shared_ptr<AVPacket>>   packets_;
+
+       std::shared_ptr<SwrContext>                             swr_                            {
+                                                                                                                                       swr_alloc_set_opts(
+                                                                                                                                                       nullptr,
+                                                                                                                                                       codec_context_->channel_layout
+                                                                                                                                                                       ? codec_context_->channel_layout
+                                                                                                                                                                       : av_get_default_channel_layout(codec_context_->channels),
+                                                                                                                                                       AV_SAMPLE_FMT_S32,
+                                                                                                                                                       out_samplerate_,
+                                                                                                                                                       codec_context_->channel_layout
+                                                                                                                                                                       ? codec_context_->channel_layout
+                                                                                                                                                                       : av_get_default_channel_layout(codec_context_->channels),
+                                                                                                                                                       codec_context_->sample_fmt,
+                                                                                                                                                       codec_context_->sample_rate,
+                                                                                                                                                       0,
+                                                                                                                                                       nullptr),
+                                                                                                                                       [](SwrContext* p)
+                                                                                                                                       {
+                                                                                                                                               swr_free(&p);
+                                                                                                                                       }
+                                                                                                                               };
+
+public:
+       explicit implementation(const spl::shared_ptr<AVFormatContext>& context, int out_samplerate)
+               : codec_context_(open_codec(*context, AVMEDIA_TYPE_AUDIO, index_, false))
+               , out_samplerate_(out_samplerate)
+               , buffer_(10 * out_samplerate_ * codec_context_->channels) // 10 seconds of audio
+       {       
+               if(!swr_)
+                       BOOST_THROW_EXCEPTION(bad_alloc());
+               
+               THROW_ON_ERROR2(swr_init(swr_.get()), "[audio_decoder]");
+
+               codec_context_->refcounted_frames = 1;
+       }
+
+       void push(const std::shared_ptr<AVPacket>& packet)
+       {                       
+               if(!packet)
+                       return;
+
+               if(packet->stream_index == index_ || packet->data == nullptr)
+                       packets_.push(spl::make_shared_ptr(packet));
+       }       
+       
+       std::shared_ptr<core::mutable_audio_buffer> poll()
+       {
+               if(packets_.empty())
+                       return nullptr;
+                               
+               auto packet = packets_.front();
+
+               if(packet->data == nullptr)
+               {
+                       packets_.pop();
+                       avcodec_flush_buffers(codec_context_.get());
+                       return flush_audio();
+               }
+
+               auto audio = decode(*packet);
+
+               if(packet->size == 0)                                   
+                       packets_.pop();
+
+               return audio;
+       }
+
+       std::shared_ptr<core::mutable_audio_buffer> decode(AVPacket& pkt)
+       {
+               auto decoded_frame = create_frame();
+
+               int got_frame = 0;
+               auto len = THROW_ON_ERROR2(avcodec_decode_audio4(codec_context_.get(), decoded_frame.get(), &got_frame, &pkt), "[audio_decoder]");
+
+               if (len == 0)
+               {
+                       pkt.size = 0;
+                       return nullptr;
+               }
+
+               pkt.data += len;
+               pkt.size -= len;
+
+               if (!got_frame)
+                       return nullptr;
+
+               const uint8_t **in = const_cast<const uint8_t**>(decoded_frame->extended_data);
+               uint8_t* out[] = { reinterpret_cast<uint8_t*>(buffer_.data()) };
+
+               const auto channel_samples = swr_convert(
+                               swr_.get(),
+                               out,
+                               static_cast<int>(buffer_.size()) / codec_context_->channels,
+                               in,
+                               decoded_frame->nb_samples);
+
+               return std::make_shared<core::mutable_audio_buffer>(
+                               buffer_.begin(),
+                               buffer_.begin() + channel_samples * decoded_frame->channels);
+       }
+
+       bool ready() const
+       {
+               return packets_.size() > 10;
+       }
+
+       std::wstring print() const
+       {               
+               return L"[audio-decoder] " + u16(codec_context_->codec->long_name);
+       }
+};
+
+audio_decoder::audio_decoder(const spl::shared_ptr<AVFormatContext>& context, int out_samplerate) : impl_(new implementation(context, out_samplerate)){}
+void audio_decoder::push(const std::shared_ptr<AVPacket>& packet){impl_->push(packet);}
+bool audio_decoder::ready() const{return impl_->ready();}
+std::shared_ptr<core::mutable_audio_buffer> audio_decoder::poll() { return impl_->poll(); }
+int    audio_decoder::num_channels() const { return impl_->codec_context_->channels; }
+uint64_t audio_decoder::ffmpeg_channel_layout() const { return impl_->codec_context_->channel_layout; }
+std::wstring audio_decoder::print() const{return impl_->print();}
+
+}}