]> git.sesse.net Git - casparcg/blobdiff - modules/ffmpeg/producer/audio/audio_decoder.cpp
2.0. audio:
[casparcg] / modules / ffmpeg / producer / audio / audio_decoder.cpp
index fee090509ae1f51ab9219c014bfdc71a6bf12bac..a71c672c25271f23c474f59fb00aecefbc2e535e 100644 (file)
 \r
 #include "audio_decoder.h"\r
 \r
-#include <tbb/task_group.h>\r
+#include "audio_resampler.h"\r
+\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
@@ -29,8 +37,6 @@
 #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
@@ -45,132 +51,119 @@ struct audio_decoder::implementation : boost::noncopyable
        std::shared_ptr<AVCodecContext>                                                         codec_context_;         \r
        const core::video_format_desc                                                           format_desc_;\r
        int                                                                                                                     index_;\r
-       std::vector<int8_t, tbb::cache_aligned_allocator<int8_t>>       buffer1_;               // avcodec_decode_audio3 needs 4 byte alignment\r
-       std::vector<int8_t, tbb::cache_aligned_allocator<int8_t>>       buffer2_;               // avcodec_decode_audio3 needs 4 byte alignment\r
-       std::vector<int16_t, tbb::cache_aligned_allocator<int16_t>>     audio_samples_;         // avcodec_decode_audio3 needs 4 byte alignment\r
+       std::unique_ptr<audio_resampler>                                                        resampler_;\r
+\r
+       std::vector<int8_t,  tbb::cache_aligned_allocator<int8_t>>      buffer1_;\r
+\r
        std::queue<std::shared_ptr<AVPacket>>                                           packets_;\r
-       std::shared_ptr<ReSampleContext>                                                        resampler_;\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
+       explicit implementation(const safe_ptr<AVFormatContext>& context, const core::video_format_desc& format_desc) \r
                : format_desc_(format_desc)     \r
-       {                          \r
-               AVCodec* dec;\r
-               index_ = av_find_best_stream(context.get(), AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);\r
+               , nb_frames_(0)\r
+       {                               \r
+               try\r
+               {\r
+                       AVCodec* dec;\r
+                       index_ = THROW_ON_ERROR2(av_find_best_stream(context.get(), AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0), "[audio_decoder]");\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
+                       THROW_ON_ERROR2(avcodec_open(context->streams[index_]->codec, dec), "[audio_decoder]");\r
+                       \r
+                       codec_context_.reset(context->streams[index_]->codec, avcodec_close);\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
+                       buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\r
 \r
-                       CASPAR_LOG(warning) << L" Invalid audio format.";\r
+                       resampler_.reset(new audio_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
+               }\r
+               catch(...)\r
+               {\r
+                       index_ = THROW_ON_ERROR2(av_find_best_stream(context.get(), AVMEDIA_TYPE_VIDEO, -1, -1, nullptr, 0), "[audio_decoder]");\r
 \r
-                       if(resampler)\r
-                               resampler_.reset(resampler, audio_resample_close);\r
-                       else\r
-                               codec_context_ = nullptr;\r
-               }               \r
+                       CASPAR_LOG_CURRENT_EXCEPTION();\r
+                       CASPAR_LOG(warning) << "[audio_decoder] Failed to open audio-stream. Running without audio.";                   \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::vector<int16_t>> poll()\r
+       std::vector<std::shared_ptr<core::audio_buffer>> poll()\r
        {\r
-               std::vector<std::vector<int16_t>> result;\r
+               std::vector<std::shared_ptr<core::audio_buffer>> result;\r
+\r
+               if(packets_.empty())\r
+                       return result;\r
 \r
                if(!codec_context_)\r
-                       result.push_back(std::vector<int16_t>(format_desc_.audio_samples_per_frame, 0));\r
-               else if(!packets_.empty())\r
+                       return empty_poll();\r
+               \r
+               auto packet = packets_.front();\r
+\r
+               if(packet)              \r
                {\r
-                       decode(packets_.front());\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
 \r
-                       while(audio_samples_.size() > format_desc_.audio_samples_per_frame)\r
-                       {\r
-                               const auto begin = audio_samples_.begin();\r
-                               const auto end   = audio_samples_.begin() + format_desc_.audio_samples_per_frame;\r
+               return result;\r
+       }\r
 \r
-                               result.push_back(std::vector<int16_t>(begin, end));\r
-                               audio_samples_.erase(begin, end);\r
-                       }\r
-               }\r
+       std::vector<std::shared_ptr<core::audio_buffer>> empty_poll()\r
+       {\r
+               auto packet = packets_.front();\r
+               packets_.pop();\r
 \r
-               return result;\r
+               if(!packet)                     \r
+                       return boost::assign::list_of(nullptr);\r
+               \r
+               return boost::assign::list_of(std::make_shared<core::audio_buffer>(format_desc_.audio_samples_per_frame, 0));   \r
        }\r
 \r
-       void decode(const std::shared_ptr<AVPacket>& packet)\r
-       {                                                                                       \r
-               if(!packet) // eof\r
-               {\r
-                       auto truncate = audio_samples_.size() % format_desc_.audio_samples_per_frame;\r
-                       if(truncate > 0)\r
-                       {\r
-                               audio_samples_.resize(audio_samples_.size() - truncate); \r
-                               CASPAR_LOG(info) << L"Truncating " << truncate << L" audio-samples."; \r
-                       }\r
-                       avcodec_flush_buffers(codec_context_.get());\r
-               }\r
-               else\r
-               {\r
-                       buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2, 0);\r
-                       int written_bytes = buffer1_.size() - FF_INPUT_BUFFER_PADDING_SIZE;\r
-                       // TODO: Packet might contain multiple frames\r
-                       const int errn = avcodec_decode_audio3(codec_context_.get(), reinterpret_cast<int16_t*>(buffer1_.data()), &written_bytes, packet.get());\r
-                       if(errn < 0)\r
-                       {       \r
-                               BOOST_THROW_EXCEPTION(\r
-                                       invalid_operation() <<\r
-                                       boost::errinfo_api_function("avcodec_decode_audio2") <<\r
-                                       boost::errinfo_errno(AVUNERROR(errn)));\r
-                       }\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)); \r
-                               buffer2_.resize(ret);\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
-                       audio_samples_.insert(audio_samples_.end(), samples, samples + n_samples);      \r
-               }\r
+       std::shared_ptr<core::audio_buffer> decode(AVPacket& pkt)\r
+       {               \r
+               buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\r
+               int written_bytes = buffer1_.size() - FF_INPUT_BUFFER_PADDING_SIZE;\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
+               pkt.data += ret;\r
+                       \r
+               buffer1_.resize(written_bytes);\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
+               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_.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
+audio_decoder::audio_decoder(const safe_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::vector<int16_t>> audio_decoder::poll(){return impl_->poll();}\r
+std::vector<std::shared_ptr<core::audio_buffer>> 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