AVI dumper: Do high-quality audio resampling in dedicated thread
Gives faster dumping in mode 4 for quadcore.
This commit is contained in:
Ilari Liusvaara
parent
8e7e6cd168
commit
c49f305892
1 changed files with 133 additions and 56 deletions
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@ -25,6 +25,9 @@
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namespace
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{
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class avi_avsnoop;
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avi_avsnoop* vid_dumper;
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uint32_t rates[] = {8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000,
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128000, 176400, 192000};
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@ -111,6 +114,24 @@ namespace
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uint32_t max_frames;
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};
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struct resample_worker : public worker_thread
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{
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resample_worker(double _ratio, uint32_t _nch);
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~resample_worker();
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void entry();
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void sendblock(short* block, size_t frames);
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void sendend();
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private:
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std::vector<short> buffers;
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std::vector<float> buffers2;
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std::vector<float> buffers3;
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std::vector<short> buffers4;
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size_t bufused;
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double ratio;
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uint32_t nch;
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void* resampler;
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};
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struct avi_worker : public worker_thread
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{
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avi_worker(const struct avi_info& info);
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@ -134,7 +155,6 @@ namespace
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#define WORKFLAG_FLUSH 2
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#define WORKFLAG_END 4
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avi_worker::avi_worker(const struct avi_info& info)
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: aviout(info.prefix, *info.vcodec, *info.acodec, info.sample_rate, info.audio_chans)
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{
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@ -212,6 +232,64 @@ namespace
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}
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}
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resample_worker::resample_worker(double _ratio, uint32_t _nch)
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{
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ratio = _ratio;
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nch = _nch;
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buffers.resize(RESAMPLE_BUFFER * nch);
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buffers2.resize(RESAMPLE_BUFFER * nch);
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buffers3.resize((RESAMPLE_BUFFER * nch * ratio) + 128 * nch);
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buffers4.resize((RESAMPLE_BUFFER * nch * ratio) + 128 * nch);
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bufused = 0;
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#ifdef WITH_SECRET_RABBIT_CODE
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int errc = 0;
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resampler = src_new(SRC_SINC_BEST_QUALITY, nch, &errc);
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if(errc)
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throw std::runtime_error(std::string("Error initing libsamplerate: ") +
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src_strerror(errc));
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#else
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throw std::runtime_error("HQ sample rate conversion not available");
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#endif
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fire();
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}
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resample_worker::~resample_worker()
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{
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#ifdef WITH_SECRET_RABBIT_CODE
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src_delete((SRC_STATE*)resampler);
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#endif
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}
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void resample_worker::sendend()
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{
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rethrow();
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set_workflag(WORKFLAG_END);
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request_quit();
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}
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void resample_worker::sendblock(short* block, size_t frames)
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{
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again:
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rethrow();
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wait_busy();
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if(bufused + frames < RESAMPLE_BUFFER) {
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memcpy(&buffers[bufused * nch], block, 2 * nch * frames);
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bufused += frames;
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block += (frames * nch);
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frames = 0;
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} else if(bufused < RESAMPLE_BUFFER) {
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size_t processable = RESAMPLE_BUFFER - bufused;
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memcpy(&buffers[bufused * nch], block, 2 * nch * processable);
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block += (processable * nch);
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frames -= processable;
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bufused = RESAMPLE_BUFFER;
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}
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set_busy();
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set_workflag(WORKFLAG_QUEUE_FRAME);
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if(frames > 0)
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goto again;
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}
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void waitfn();
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class avi_avsnoop : public information_dispatch
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@ -230,32 +308,19 @@ namespace
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soundrate.second, 2);
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dcounter = 0;
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have_dumped_frame = false;
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resampler = NULL;
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resampler_w = NULL;
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if(soundrate_setting == 4) {
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double ratio = 1.0 * audio_record_rate * soundrate.second / soundrate.first;
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sbuffer.resize(RESAMPLE_BUFFER * chans);
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sbuffer2.resize(RESAMPLE_BUFFER * chans);
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fbuffer.resize((RESAMPLE_BUFFER * ratio + 128) * chans + 128);
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fbuffer2.resize((RESAMPLE_BUFFER * ratio + 128) * chans + 128);
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sbuffer_fill = 0;
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#ifdef WITH_SECRET_RABBIT_CODE
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int errc = 0;
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resampler = src_new(SRC_SINC_BEST_QUALITY, info.audio_chans, &errc);
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if(errc)
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throw std::runtime_error(std::string("Error initing libsamplerate: ") +
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src_strerror(errc));
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#else
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throw std::runtime_error("HQ sample rate conversion not available");
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#endif
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sbuffer.resize(RESAMPLE_BUFFER * chans);
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resampler_w = new resample_worker(ratio, chans);
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}
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}
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~avi_avsnoop() throw()
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{
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#ifdef WITH_SECRET_RABBIT_CODE
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if(resampler)
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src_delete((SRC_STATE*)resampler);
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#endif
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if(resampler_w)
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delete resampler_w;
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delete worker;
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delete soxdumper;
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}
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@ -275,12 +340,15 @@ namespace
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void on_sample(short l, short r)
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{
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if(resampler) {
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if(resampler_w) {
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if(!have_dumped_frame)
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return;
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sbuffer[sbuffer_fill++] = l;
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sbuffer[sbuffer_fill++] = r;
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forward_samples(false);
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if(sbuffer_fill == sbuffer.size()) {
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resampler_w->sendblock(&sbuffer[0], sbuffer_fill / chans);
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sbuffer_fill = 0;
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}
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soxdumper->sample(l, r);
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return;
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}
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@ -301,7 +369,8 @@ namespace
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void on_dump_end()
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{
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if(worker) {
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forward_samples(true);
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if(resampler_w)
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resampler_w->sendend();
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worker->request_quit();
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}
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if(soxdumper)
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@ -317,52 +386,19 @@ namespace
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return true;
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}
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avi_worker* worker;
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resample_worker* resampler_w;
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private:
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void forward_samples(bool eos)
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{
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if(!eos && sbuffer_fill < sbuffer.size())
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return;
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#ifdef WITH_SECRET_RABBIT_CODE
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double ratio = 1.0 * audio_record_rate * soundrate.second / soundrate.first;
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SRC_DATA block;
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src_short_to_float_array(&sbuffer[0], &sbuffer2[0], sbuffer_fill);
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block.data_in = &sbuffer2[0];
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block.data_out = &fbuffer2[0];
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block.input_frames = sbuffer_fill / chans;
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block.input_frames_used = 0;
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block.output_frames = fbuffer2.size() / chans;
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block.output_frames_gen = 0;
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block.end_of_input = eos ? 1 : 0;
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block.src_ratio = ratio;
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int errc = src_process((SRC_STATE*)resampler, &block);
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if(errc)
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throw std::runtime_error(std::string("Error using libsamplerate: ") +
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src_strerror(errc));
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src_float_to_short_array(&fbuffer2[0], &fbuffer[0], block.output_frames_gen * chans);
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worker->queue_audio(&fbuffer[0], block.output_frames_gen * chans);
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if(block.input_frames_used * chans < sbuffer_fill)
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memmove(&sbuffer[0], &sbuffer[block.output_frames_gen * chans], sbuffer_fill -
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block.input_frames_used * chans);
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sbuffer_fill -= block.input_frames_used * chans;
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#endif
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}
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sox_dumper* soxdumper;
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screen<false> dscr;
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unsigned dcounter;
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bool have_dumped_frame;
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std::pair<uint32_t, uint32_t> soundrate;
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uint32_t audio_record_rate;
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void* resampler;
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std::vector<short> sbuffer;
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std::vector<float> sbuffer2;
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std::vector<float> fbuffer2;
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std::vector<short> fbuffer;
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size_t sbuffer_fill;
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uint32_t chans;
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};
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avi_avsnoop* vid_dumper;
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void waitfn()
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{
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vid_dumper->worker->wait_busy();
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@ -454,4 +490,45 @@ namespace
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adv_avi_dumper::~adv_avi_dumper() throw()
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{
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}
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void resample_worker::entry()
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{
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while(1) {
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wait_workflag();
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uint32_t work = clear_workflag(~WORKFLAG_QUIT_REQUEST);
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if(work & (WORKFLAG_QUEUE_FRAME | WORKFLAG_END)) {
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#ifdef WITH_SECRET_RABBIT_CODE
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again:
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SRC_DATA block;
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src_short_to_float_array(&buffers[0], &buffers2[0], bufused * nch);
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block.data_in = &buffers2[0];
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block.data_out = &buffers3[0];
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block.input_frames = bufused;
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block.input_frames_used = 0;
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block.output_frames = buffers3.size() / nch;
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block.output_frames_gen = 0;
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block.end_of_input = (work & WORKFLAG_END) ? 1 : 0;
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block.src_ratio = ratio;
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int errc = src_process((SRC_STATE*)resampler, &block);
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if(errc)
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throw std::runtime_error(std::string("Error using libsamplerate: ") +
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src_strerror(errc));
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src_float_to_short_array(&buffers3[0], &buffers4[0], block.output_frames_gen * nch);
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vid_dumper->worker->queue_audio(&buffers4[0], block.output_frames_gen * nch);
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if(block.input_frames_used < bufused)
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memmove(&buffers[0], &buffers[block.output_frames_gen * nch], (bufused -
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block.input_frames_used) * nch);
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bufused -= block.input_frames_used;
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if(block.output_frames_gen > 0 && work & WORKFLAG_END)
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goto again; //Try again to get all the samples.
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#endif
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clear_workflag(WORKFLAG_END | WORKFLAG_FLUSH | WORKFLAG_QUEUE_FRAME);
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clear_busy();
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if(work & WORKFLAG_END)
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return;
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}
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if(work == WORKFLAG_QUIT_REQUEST)
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break;
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}
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}
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}
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