/*\r
-* copyright (c) 2010 Sveriges Television AB <info@casparcg.com>\r
+* Copyright (c) 2011 Sveriges Television AB <info@casparcg.com>\r
*\r
-* This file is part of CasparCG.\r
+* This file is part of CasparCG (www.casparcg.com).\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
+* 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
+* 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
+* Author: Robert Nagy, ronag89@gmail.com\r
*/\r
+\r
#include "../../stdafx.h"\r
\r
#include "audio_decoder.h"\r
\r
-#include <tbb/task_group.h>\r
+#include "audio_resampler.h"\r
+\r
+#include "../util/util.h"\r
+#include "../../ffmpeg_error.h"\r
+\r
+#include <core/video_format.h>\r
+\r
+#include <tbb/cache_aligned_allocator.h>\r
+\r
+#include <queue>\r
\r
#if defined(_MSC_VER)\r
#pragma warning (push)\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
#pragma warning (pop)\r
#endif\r
\r
-namespace caspar {\r
+namespace caspar { namespace ffmpeg {\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
+ const safe_ptr<AVCodecContext> codec_context_; \r
+ const core::video_format_desc format_desc_;\r
+\r
+ audio_resampler 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
+ std::queue<safe_ptr<AVPacket>> packets_;\r
+\r
+ const int64_t nb_frames_;\r
+ tbb::atomic<size_t> file_frame_number_;\r
public:\r
- explicit implementation(const s
td::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc)
\r+ explicit implementation(const s
afe_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
+ , codec_context_(open_codec(*context, AVMEDIA_TYPE_AUDIO, index_))\r
+ , resampler_(format_desc.audio_channels, codec_context_->channels,\r
+ format_desc.audio_sample_rate, codec_context_->sample_rate,\r
+ AV_SAMPLE_FMT_S32, codec_context_->sample_fmt)\r
+ , buffer1_(AVCODEC_MAX_AUDIO_FRAME_SIZE*2)\r
+ , nb_frames_(0)//context->streams[index_]->nb_frames)\r
+ { \r
+ file_frame_number_ = 0;\r
+ CASPAR_LOG(debug) << "[audio_decoder] " << context->streams[index_]->codec->codec->long_name; \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
+ if(!packet)\r
return;\r
\r
- packets_.push(packet);\r
+ if(packet->stream_index == index_ || packet->data == nullptr)\r
+ packets_.push(make_safe_ptr(packet));\r
} \r
\r
- std::vector<std::shared_ptr<std::vector<int16_t>>> poll()\r
+ std::shared_ptr<core::audio_buffer> 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 = packets_.front();\r
-\r
- if(packet) \r
- {\r
- result.push_back(decode(*packet));\r
- if(packet->size == 0) \r
- packets_.pop();\r
- }\r
- else \r
- { \r
- avcodec_flush_buffers(codec_context_.get());\r
- result.push_back(nullptr);\r
- packets_.pop();\r
- }\r
+ if(packets_.empty())\r
+ return nullptr;\r
+ \r
+ auto packet = packets_.front();\r
+\r
+ if(packet->data == nullptr)\r
+ {\r
+ packets_.pop();\r
+ file_frame_number_ = static_cast<size_t>(packet->pos);\r
+ avcodec_flush_buffers(codec_context_.get());\r
+ return flush_audio();\r
}\r
\r
- return result;\r
+ auto audio = decode(*packet);\r
+\r
+ if(packet->size == 0) \r
+ packets_.pop();\r
+\r
+ return audio;\r
}\r
\r
- std::shared_ptr<std::vector<int16_t>> decode(AVPacket& pkt)\r
+ std::shared_ptr<core::audio_buffer> decode(AVPacket& pkt)\r
{ \r
- buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2, 0);\r
+ buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\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
+ int ret = THROW_ON_ERROR2(avcodec_decode_audio3(codec_context_.get(), reinterpret_cast<int16_t*>(buffer1_.data()), &written_bytes, &pkt), "[audio_decoder]");\r
\r
// There might be several frames in one packet.\r
pkt.size -= 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
+ buffer1_ = resampler_.resample(std::move(buffer1_));\r
+ \r
+ const auto n_samples = buffer1_.size() / av_get_bytes_per_sample(AV_SAMPLE_FMT_S32);\r
+ const auto samples = reinterpret_cast<int32_t*>(buffer1_.data());\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
+ ++file_frame_number_;\r
\r
- return std::make_shared<std::vector<int16_t>>(samples, samples + n_samples);\r
+ return std::make_shared<core::audio_buffer>(samples, samples + n_samples);\r
}\r
\r
bool ready() const\r
{\r
- return !codec_context_ || !packets_.empty();\r
+ return packets_.size() > 10;\r
+ }\r
+\r
+ uint32_t nb_frames() const\r
+ {\r
+ return 0;//std::max<int64_t>(nb_frames_, file_frame_number_);\r
}\r
};\r
\r
-audio_decoder::audio_decoder(const s
td::shared_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) : impl_(new implementation(context, format_desc)){}
\r+audio_decoder::audio_decoder(const s
afe_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
+std::shared_ptr<core::audio_buffer> audio_decoder::poll(){return impl_->poll();}\r
+uint32_t audio_decoder::nb_frames() const{return impl_->nb_frames();}\r
+uint32_t audio_decoder::file_frame_number() const{return impl_->file_frame_number_;}\r
+\r
+}}
\ No newline at end of file
projects / casparcg / blobdiff