\r
#include "audio_decoder.h"\r
\r
+#include "audio_resampler.h"\r
+\r
#include "../../ffmpeg_error.h"\r
\r
#include <core/video_format.h>\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
+ std::unique_ptr<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
+\r
std::queue<std::shared_ptr<AVPacket>> packets_;\r
\r
int64_t nb_frames_;\r
\r
buffer1_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\r
\r
- if(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
- buffer2_.resize(AVCODEC_MAX_AUDIO_FRAME_SIZE*2);\r
-\r
- CASPAR_LOG(warning) << L"Invalid audio format. Resampling." <<\r
- L" sample_rate:" << static_cast<int>(codec_context_->sample_rate) <<\r
- L" audio_channels:" << static_cast<int>(codec_context_->channels) <<\r
- L" sample_fmt:" << static_cast<int>(codec_context_->sample_fmt);\r
-\r
- if(resampler)\r
- resampler_.reset(resampler, audio_resample_close);\r
- else\r
- codec_context_ = nullptr;\r
- } \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
packets_.push(packet);\r
} \r
\r
- std::vector<std::shared_ptr<std::vector<int16_t>>> poll()\r
+ std::vector<std::shared_ptr<core::audio_buffer>> poll()\r
{\r
- std::vector<std::shared_ptr<std::vector<int16_t>>> result;\r
+ std::vector<std::shared_ptr<core::audio_buffer>> result;\r
\r
if(packets_.empty())\r
return result;\r
return result;\r
}\r
\r
- std::vector<std::shared_ptr<std::vector<int16_t>>> empty_poll()\r
+ std::vector<std::shared_ptr<core::audio_buffer>> empty_poll()\r
{\r
auto packet = packets_.front();\r
packets_.pop();\r
if(!packet) \r
return boost::assign::list_of(nullptr);\r
\r
- return boost::assign::list_of(std::make_shared<std::vector<int16_t>>(format_desc_.audio_samples_per_frame, 0)); \r
+ return boost::assign::list_of(std::make_shared<core::audio_buffer>(format_desc_.audio_samples_per_frame, 0)); \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);\r
int written_bytes = buffer1_.size() - FF_INPUT_BUFFER_PADDING_SIZE;\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
pkt.data += ret;\r
\r
- if(resampler_)\r
- {\r
- auto ret = audio_resample(resampler_.get(),\r
- reinterpret_cast<short*>(buffer2_.data()), \r
- reinterpret_cast<short*>(buffer1_.data()), \r
- written_bytes / (av_get_bytes_per_sample(codec_context_->sample_fmt) * codec_context_->channels)); \r
- written_bytes = ret * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * format_desc_.audio_channels;\r
- std::swap(buffer1_, buffer2_);\r
- }\r
+ buffer1_.resize(written_bytes);\r
\r
- const auto n_samples = written_bytes / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);\r
- const auto samples = reinterpret_cast<int16_t*>(buffer1_.data());\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<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
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::shared_ptr<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
+\r
+}}
\ No newline at end of file
projects / casparcg / blobdiff