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-rw-r--r--rotord/libavaudioloader.cpp231
1 files changed, 0 insertions, 231 deletions
diff --git a/rotord/libavaudioloader.cpp b/rotord/libavaudioloader.cpp
deleted file mode 100644
index a634740..0000000
--- a/rotord/libavaudioloader.cpp
+++ /dev/null
@@ -1,231 +0,0 @@
-#include "libavaudioloader.h"
-
-bool libav::Audioloader::setup(const std::string &filename){
-
- av_register_all();
-
- frame = avcodec_alloc_frame();
- if (!frame)
- {
- std::cout << "Error allocating the frame" << std::endl;
- return false;
- }
-
- formatContext = NULL;
- if (avformat_open_input(&formatContext, filename.c_str(), NULL, NULL) != 0)
- {
- av_free(frame);
- std::cout << "Error opening the file" << std::endl;
- return false;
- }
-
- if (avformat_find_stream_info(formatContext, NULL) < 0)
- {
- av_free(frame);
- avformat_close_input(&formatContext);
- std::cout << "Error finding the stream info" << std::endl;
- return false;
- }
-
- audioStream = NULL;
- for (unsigned int i = 0; i < formatContext->nb_streams; ++i)
- {
- if (formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
- {
- audioStream = formatContext->streams[i];
- break;
- }
- }
-
- if (audioStream == NULL)
- {
- av_free(frame);
- avformat_close_input(&formatContext);
- std::cout << "Could not find any audio stream in the file" << std::endl;
- return false;
- }
-
- codecContext = audioStream->codec;
-
- codecContext->codec = avcodec_find_decoder(codecContext->codec_id);
- if (codecContext->codec == NULL)
- {
- av_free(frame);
- avformat_close_input(&formatContext);
- std::cout << "Couldn't find a proper decoder" << std::endl;
- return false;
- }
- else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
- {
- av_free(frame);
- avformat_close_input(&formatContext);
- std::cout << "Couldn't open the context with the decoder" << std::endl;
- return false;
- }
-
- av_dump_format(formatContext, 0, 0, false); //avformat.h line 1256
- int samples = ((formatContext->duration + 5000)*codecContext->sample_rate)/AV_TIME_BASE;
-
- std::cout << "This stream has " << codecContext->channels << " channels, a sample rate of " << codecContext->sample_rate << "Hz and "<<samples <<" samples" << std::endl;
- std::cout << "The data is in format " <<codecContext->sample_fmt<< " (aka "<< av_get_sample_fmt_name(codecContext->sample_fmt) << ") "<<std::endl;
-
- av_init_packet(&packet);
- //sample_processed=0;
- ready=true;
- return true;
- }
- AVFrame* libav::Audioloader::get_frame() {
-
- if (!ready) return nullptr;
-
- int frameFinished = 0;
- while (!frameFinished) {
- int ret=av_read_frame(formatContext, &packet);
- if (ret<0) {
- std::cerr << "finished with code "<<ret <<(ret==AVERROR_EOF?" ,EOF":"")<<std::endl;
- ready=false;
- return nullptr;
- }
- if (packet.stream_index == audioStream->index)
- {
- //int bytes =
- avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);
-
- // Some frames rely on multiple packets, so we have to make sure the frame is finished before
- // we can use it
- }
- // You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
- av_free_packet(&packet);
- }
- return frame;
-
- }
- AVPacket* libav::Audioloader::get_packet() {
-
- if (!ready) return nullptr;
-
- int ret=av_read_frame(formatContext, &packet);
- if (ret<0) {
- std::cerr << "finished with code "<<ret <<(ret==AVERROR_EOF?" ,EOF":"")<<std::endl;
- ready=false;
- return nullptr;
- }
- //if (packet.stream_index == audioStream->index)
- //{
- //int bytes =
- // avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);
-
- // Some frames rely on multiple packets, so we have to make sure the frame is finished before
- // we can use it
- //}
- // You *must* call av_free_packet() after each call to av_read_frame() or else you'll leak memory
- //av_free_packet(&packet);?????
- //}
- return &packet;
-
- }
-
-/*
- // Some codecs will cause frames to be buffered up in the decoding process. If the CODEC_CAP_DELAY flag
- // is set, there can be buffered up frames that need to be flushed, so we'll do that
- if (codecContext->codec->capabilities & CODEC_CAP_DELAY)
- {
- av_init_packet(&packet);
- // Decode all the remaining frames in the buffer, until the end is reached
- int frameFinished = 0;
- int bytes = avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);
- while (bytes >= 0 && frameFinished)
- {
- for (auto p: processors) {
- p->process_frame(frame->data[0],frame->nb_samples);
- }
- mutex.lock();
- progress=((double)sample_processed)/samples;
- mutex.unlock();
- }
- }
-
- cerr << "finished processing: "<<sample_processed << " samples of "<<samples<<", "<<((double)sample_processed*100)/samples<<"%"<< std::endl;
-*/
-
-uint16_t* libav::Audioloader::get_samples(int num){ //presumes 16bpc here
- //std::cerr << "request "<<num<<" samples: "<<(ready?"ready":"not ready")<<std::endl;
- //if(!ready) return nullptr;
- //shuffle down samples
-
- if (sample_start>0){
- for (int i=0;i<sample_end-sample_start;i++){
- for (int j=0;j<channels;j++) {
- buffer[(i*channels)+j]=buffer[((sample_start+i)*channels)+j];
- }
- }
- sample_start=sample_end-sample_start;
- }
-
- sample_end=sample_start;
- while (sample_end<num) {
- frame=get_frame();
- if (frame) {
- channels=av_frame_get_channels(frame); //will always reach here 1st
- if (((sample_end+std::max(num,frame->nb_samples))*channels)>buffer.size()){
- int m=buffer.size();
- int s=((sample_end+std::max(num,frame->nb_samples))*channels);
- buffer.reserve(s);
- std::cerr << "audioloader reserved buffer to " << s << std::endl;
- for (int i=m;i<s;i++) buffer.push_back(0);
- }
- for (int i=0;i<frame->nb_samples;i++) {
- for (int j=0;j<channels;j++) {
- //int frame->format
- //format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames, enum AVSampleFormat for audio)
- int ff=frame->format;
- //uint64_t frame->channel_layout
- //Channel layout of the audio data.
- uint64_t fcl=frame->channel_layout;
- //int frame->nb_extended_buf
- //Number of elements in extended_buf.
- int fnb=frame->nb_extended_buf;
- //int frame->decode_error_flags
- //decode error flags of the frame, set to a combination of FF_DECODE_ERROR_xxx flags if the decoder produced a frame, but there were errors during the decoding.
- int fde=frame->decode_error_flags;
-
-
- //uint16_t s=((uint16_t*) frame->buf[j]->data)[i];
- uint16_t s=((uint16_t*) frame->buf[0]->data)[j*channels+i];
- //which? must be determined by format or layout of the channels
- //ALSO some kind of HEINOUS memory leak??
- buffer[((sample_end+i)*frame->channels)+j]=s;
- //buffer[(j*frame->channels)+(sample_end+i)]= ((uint16_t*) frame->buf[j]->data)[i]; ??planar?? nope
- }
- }
- sample_end+=frame->nb_samples;
- }
- else {
- for (int i=sample_end;i<num;i++){
- for (int j=0;j<channels;j++) {
- buffer[(channels*i)+j]=0;
- }
- }
- sample_end=num;
- }
- //std::cerr<<"filling buffer to "<<((sample_end+frame->nb_samples)*frame->channels)<<std::endl;
-
-
- //avcodec_free_frame(&frame);
- }
- if (sample_end>num) {
- sample_start=num;
- }
- else {
- sample_start=0;
- }
- return (uint16_t*)(&buffer[0]);
-}
-
-bool libav::Audioloader::close() {
- av_free(frame);
- avcodec_close(codecContext);
- avformat_close_input(&formatContext);
-
- return true;
-}
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