Adding Opus
Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
Mahyar Koshkouei
151
source/main.c
151
source/main.c
@@ -17,18 +17,12 @@
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#include <string.h>
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#include "main.h"
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#include "trivial_example.h"
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#define BUFFER_SIZE 16 * 1024
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#define AUDIO_FOLDER "sdmc:/MUSIC/"
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#define CHANNEL 0x08
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// Temporary Opus defines
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#define CHANNELS 2
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#define MAX_FRAME_SIZE 6*960
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#define MAX_PACKET_SIZE (3*1276)
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#define FRAME_SIZE 960
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/* Adds extra debugging text */
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#define DEBUG 0
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@@ -141,7 +135,14 @@ int main(int argc, char **argv)
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if(file == NULL)
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err_print("Opening file failed.");
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else
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playOpus(file); // TODO: make this dynamic
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{
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int ret;
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// TODO: make this dynamic
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if((ret = convOpus(file, "sdmc:/MUSIC/out.wav")) != 0)
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playWav(file);
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printf("ret=%d\n", ret);
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}
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free(file);
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}
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@@ -348,83 +349,91 @@ out:
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return 0;
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}
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/**
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* Plays an Opus encoded music file.
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*
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* \param file File location of Opus file.
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* \return Zero if successful, else failure.
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*/
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int playOpus(const char *opus)
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#if 0
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int playOpus(const char* opus)
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{
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err_print("Here");
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FILE *file = fopen(opus, "rb");
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opus_int16 *out = malloc(MAX_FRAME_SIZE*CHANNELS);
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OpusDecoder *decoder;
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int err;
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ndspWaveBuf waveBuf[1];
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OggOpusFile *of;
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int ret;
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int output_seekable;
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FILE* outfile;
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ogg_int64_t pcm_offset;
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ogg_int64_t pcm_print_offset;
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ogg_int64_t nsamples;
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opus_int32 bitrate = 0;
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int prev_li;
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err_print("Here");
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decoder = opus_decoder_create(48000, CHANNELS, &err);
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if(err < 0)
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printf("Size: %u\n", sizeof(of));
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of = op_open_file(opus, &ret);
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if(of == NULL)
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{
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fprintf(stderr, "failed to set bitrate: %s\n", opus_strerror(err));
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return 1;
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fprintf(stderr,"Failed to open file '%s': %i\n", opus, ret);
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return -1;
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}
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if(file == NULL)
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{
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err_print("Opening file failed.");
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return 1;
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outfile = fopen("sdmc:/MUSIC/out.wav", "wb+");
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if(outfile == NULL){
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fprintf(stderr,"Failed to open output file : %i\n",ret);
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return EXIT_FAILURE;
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}
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err_print("Here");
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ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
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ndspChnSetRate(CHANNEL, 48000);
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ndspChnSetFormat(CHANNEL, NDSP_FORMAT_STEREO_PCM16);
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memset(waveBuf, 0, sizeof(waveBuf));
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//buffer1 = (u32*) linearAlloc(BUFFER_SIZE);
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waveBuf[0].nsamples = MAX_FRAME_SIZE*CHANNELS*2;
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pcm_offset = op_pcm_tell(of);
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if(pcm_offset != 0){
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fprintf(stderr, "Non-zero starting PCM offset: %li\n", (long)pcm_offset);
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}
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err_print("Here");
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while(1)
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{
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int i;
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unsigned char* pcm_bytes = (unsigned char*)linearAlloc(BUFFER_SIZE);
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int frame_size;
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printf("pcm_offset: %li\n", pcm_offset);
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pcm_print_offset = pcm_offset - 48000;
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/* Read a 16 bits/sample audio frame. */
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fread(pcm_bytes, 1, BUFFER_SIZE, file);
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for(;;){
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ogg_int64_t next_pcm_offset;
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opus_int16* pcm = (opus_int16*)malloc(120*48*2*sizeof(opus_int16));
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unsigned char* out = (unsigned char*)malloc(120*48*2*2*sizeof(unsigned char));
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int li;
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int si;
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static int count = 0;
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if (feof(file))
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break;
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err_print("Here");
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/* Decode the data. In this example, frame_size will be constant because
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the encoder is using a constant frame size. However, that may not
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be the case for all encoders, so the decoder must always check
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the frame size returned. */
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frame_size = opus_decode(decoder, pcm_bytes,
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sizeof(pcm_bytes)/sizeof(unsigned char), out, MAX_FRAME_SIZE, 0);
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err_print("Here");
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if(frame_size < 0)
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ret = op_read_stereo(of, pcm, sizeof(pcm)/sizeof(*pcm));
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if(ret < 0)
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{
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fprintf(stderr, "decoder failed: %s\n", opus_strerror(frame_size));
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goto out;
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fprintf(stderr, "\nError decoding '%s': %i\n", opus, ret);
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ret = EXIT_FAILURE;
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break;
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}
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printf("\rCount: %d", count++);
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waveBuf[0].data_vaddr = &pcm_bytes;
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next_pcm_offset=op_pcm_tell(of);
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if(pcm_offset+ret!=next_pcm_offset){
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fprintf(stderr,"\nPCM offset gap! %li+%i!=%li\n",
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(long)pcm_offset,ret,(long)next_pcm_offset);
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}
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pcm_offset=next_pcm_offset;
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if(ret<=0){
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ret=EXIT_SUCCESS;
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break;
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}
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/*Ensure the data is little-endian before writing it out.*/
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for(si=0;si<2*ret;si++){
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out[2*si+0]=(unsigned char)(pcm[si]&0xFF);
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out[2*si+1]=(unsigned char)(pcm[si]>>8&0xFF);
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}
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if(!fwrite(out,sizeof(*out)*4*ret,1,outfile)){
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fprintf(stderr,"\nError writing decoded audio data: %s\n",
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strerror(errno));
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ret=EXIT_FAILURE;
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break;
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}
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nsamples+=ret;
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prev_li=li;
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while(waveBuf[0].status != NDSP_WBUF_DONE)
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{}
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free(pcm_bytes);
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free(pcm);
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}
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out:
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/*Destroy the encoder state*/
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opus_decoder_destroy(decoder);
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fclose(file);
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free(out);
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return EXIT_SUCCESS;
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if(ret==EXIT_SUCCESS){
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fprintf(stderr,"\nDone: played ");
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fprintf(stderr," (%li samples @ 48 kHz).\n",(long)nsamples);
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}
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fclose(outfile);
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op_free(of);
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return ret;
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}
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#endif
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