wrestling with libavcodec

1 files changed, 93 insertions, 0 deletions

diff --git a/avcodec_audio_example1.c b/avcodec_audio_example1.c
new file mode 100644
index 0000000..b296c00
--- /dev/null
+++ b/avcodec_audio_example1.c
@@ -0,0 +1,93 @@
+bool Render_context::load_audio(string &filename){
+	//load audio into memory and create thumbnail
+	AVCodec *codec;
+	AVCodecContext *c= NULL;
+	int len;
+	FILE *f;
+	uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+	AVPacket avpkt;
+	AVFrame *decoded_frame = NULL;
+
+	av_init_packet(&avpkt);
+
+	// http://stackoverflow.com/questions/9799560/decode-audio-using-libavcodec-and-play-using-libao
+	
+	/* find the mpeg audio decoder */
+	codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
+	if (!codec) {
+		fprintf(stderr, "Codec not found\n");
+		return false;
+	}
+
+	c = avcodec_alloc_context3(codec);
+	if (!c) {
+		fprintf(stderr, "Could not allocate audio codec context\n");
+		return false;
+	}
+
+	/* open it */
+	if (avcodec_open2(c, codec, NULL) < 0) {
+		fprintf(stderr, "Could not open codec\n");
+		return false;
+	}
+
+	f = fopen(filename.c_str(), "rb");
+	if (!f) {
+		fprintf(stderr, "Could not open %s\n", filename.c_str());
+		return false;
+	}
+
+	/* decode until eof */
+	avpkt.data = inbuf;
+	avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+	while (avpkt.size > 0) {
+		int got_frame = 0;
+
+		if (!decoded_frame) {
+			if (!(decoded_frame = avcodec_alloc_frame())) {
+				printf("Could not allocate audio frame\n");
+			return false;
+			}
+		} else
+		avcodec_get_frame_defaults(decoded_frame);
+
+		len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+		if (len < 0) {
+			fprintf(stderr, "Error while decoding\n");
+			return false;
+		}
+		if (got_frame) {
+			/* if a frame has been decoded, output it */
+			int data_size = av_samples_get_buffer_size(NULL, c->channels,decoded_frame->nb_samples,c->sample_fmt, 1);
+			//fwrite(decoded_frame->data[0], 1, data_size, outfile);
+			
+			//here is where we do something with the data
+			//is it best to buffer it or do do all processing within a loop like this?
+			
+		}
+		avpkt.size -= len;
+		avpkt.data += len;
+		avpkt.dts =
+		avpkt.pts = AV_NOPTS_VALUE;
+		if (avpkt.size < AUDIO_REFILL_THRESH) {
+			/* Refill the input buffer, to avoid trying to decode
+			* incomplete frames. Instead of this, one could also use
+			* a parser, or use a proper container format through
+			* libavformat. */
+			memmove(inbuf, avpkt.data, avpkt.size);
+			avpkt.data = inbuf;
+			len = fread(avpkt.data + avpkt.size, 1,AUDIO_INBUF_SIZE - avpkt.size, f);
+			if (len > 0)
+			avpkt.size += len;
+		}
+	}
+
+	fclose(f);
+
+	avcodec_close(c);
+	av_free(c);
+	avcodec_free_frame(&decoded_frame);
+    
+	return true;
+}
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