audio muxer ongoing

1 files changed, 133 insertions, 0 deletions

diff --git a/rotord/libavaudioloader.cpp b/rotord/libavaudioloader.cpp
new file mode 100644
index 0000000..8a2d81a
--- /dev/null
+++ b/rotord/libavaudioloader.cpp
@@ -0,0 +1,133 @@
+#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;
+
+	}
+
+/*
+		// 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;
+*/
+
+	bool libav::audioloader::close() {
+		av_free(frame);
+		avcodec_close(codecContext);
+		avformat_close_input(&formatContext);
+
+		return true;
+	}
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