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Attempting to play opus on the fly

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This doesn't work yet.

Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
Mahyar Koshkouei
2016-12-09 13:39:23 +00:00
parent 2805088a4a
commit 7e33638380
2 changed files with 63 additions and 32 deletions
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1
source/main.c
View File

@@ -183,6 +183,7 @@ int playWav(const char *wav)
goto out; goto out;
} }
// TODO: Check if this is required.
ndspSetOutputMode(NDSP_OUTPUT_STEREO); ndspSetOutputMode(NDSP_OUTPUT_STEREO);
if(file == NULL) if(file == NULL)

94
source/trivial_example.c
View File

@@ -29,7 +29,9 @@
# define fileno _fileno # define fileno _fileno
#endif #endif
#define BUFFER_SIZE 120*48*2 // Number of samples per second * channels * size of one sample
#define CHANNEL 0x08
#define BUFFER_SIZE 120*48*2*sizeof(opus_int16)*32
int convOpus(const char* in, const char *outf) int convOpus(const char* in, const char *outf)
{ {
@@ -39,20 +41,21 @@ int convOpus(const char* in, const char *outf)
int ret; int ret;
int output_seekable; int output_seekable;
ogg_int64_t nsamples = 0; ogg_int64_t nsamples = 0;
opus_int16* pcm = malloc(BUFFER_SIZE*sizeof(opus_int16)); opus_int16* pcm = malloc(120*48*2*sizeof(opus_int16));
unsigned char* out = malloc(BUFFER_SIZE*2*sizeof(unsigned char)); // Buffer to play one second.
ndspWaveBuf waveBuf[2];
of = op_open_file(in, &ret); of = op_open_file(in, &ret);
if((wav = fopen(outf, "w+")) == NULL) if((wav = fopen(outf, "w+")) == NULL)
{ {
fprintf(stderr,"Failed to open file '%s': %i %s\n",outf,errno, strerror(errno)); fprintf(stderr, "Failed to open file '%s': %i %s\n",outf,errno, strerror(errno));
return EXIT_FAILURE; return EXIT_FAILURE;
} }
if(of == NULL) if(of == NULL)
{ {
fprintf(stderr,"Failed to open file '%s': %i\n",in,ret); fprintf(stderr, "Failed to open file '%s': %i\n",in,ret);
return EXIT_FAILURE; return EXIT_FAILURE;
} }
@@ -60,27 +63,53 @@ int convOpus(const char* in, const char *outf)
if(op_seekable(of)) if(op_seekable(of))
{ {
fprintf(stderr,"Total number of links: %i\n",op_link_count(of)); fprintf(stderr, "\nTotal number of links: %i\n", op_link_count(of));
// Get PCM length of entire stream. // Get PCM length of entire stream.
duration = op_pcm_total(of, -1); duration = op_pcm_total(of, -1);
fprintf(stderr,"Total duration: "); fprintf(stderr, "Total duration: ");
fprintf(stderr," (%li samples @ 48 kHz)\n",(long)duration); fprintf(stderr, " (%li samples @ 48 kHz)\n", (long)duration);
} }
else if(output_seekable < 0) else if(output_seekable < 0)
{
fprintf(stderr,"WARNING: Neither input nor output are seekable.\n"); fprintf(stderr,"WARNING: Neither input nor output are seekable.\n");
fprintf(stderr,
"Writing non-standard WAV header with invalid chunk sizes.\n"); /* Initialise DSP stuff */
if(R_FAILED(ndspInit()))
{
printf("Initialising ndsp failed.");
goto out;
} }
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
ndspChnReset(CHANNEL);
ndspChnWaveBufClear(CHANNEL);
/* Polyphase sounds much better than linear or no interpolation */
ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
ndspChnSetRate(CHANNEL, 48000);
ndspChnSetFormat(CHANNEL, NDSP_FORMAT_STEREO_PCM16);
memset(waveBuf, 0, sizeof(waveBuf));
fprintf(stderr, "PCM %d, %d\n", sizeof(pcm), sizeof(*pcm));
for(;;) for(;;)
{ {
int si; u32 kDown;
/*Although we would generally prefer to use the float interface, WAV hidScanInput();
files with signed, 16-bit little-endian samples are far more gfxFlushBuffers();
universally supported, so that's what we output.*/ gfxSwapBuffers();
ret = op_read_stereo(of, pcm, sizeof(pcm)/sizeof(*pcm)); gspWaitForVBlank();
kDown = hidKeysDown();
if(kDown & KEY_B)
break;
/*
* Although we would generally prefer to use the float interface, WAV
* files with signed, 16-bit little-endian samples are far more
* universally supported, so that's what we output.
*/
ret = op_read_stereo(of, pcm, 120*48*2);
if(ret < 0) if(ret < 0)
{ {
@@ -95,14 +124,12 @@ int convOpus(const char* in, const char *outf)
break; break;
} }
/*Ensure the data is little-endian before writing it out.*/ nsamples += ret;
for(si=0; si<2*ret; si++)
{
out[2*si+0]=(unsigned char)(pcm[si]&0xFF);
out[2*si+1]=(unsigned char)(pcm[si]>>8&0xFF);
}
if(fwrite(out, sizeof(u32), 1, wav) == 0) fprintf(stderr, "\rRet: %d, Sample: %li", ret, (long)nsamples);
// Number of samples read * size of each sample * number of channels
if(fwrite(pcm, ret * sizeof(opus_int16) * 2, 1, wav) == 0)
{ {
fprintf(stderr, "\nError writing decoded audio data: %s\n", fprintf(stderr, "\nError writing decoded audio data: %s\n",
strerror(errno)); strerror(errno));
@@ -110,18 +137,19 @@ int convOpus(const char* in, const char *outf)
break; break;
} }
nsamples += ret; #if 0
waveBuf[0].data_vaddr = &pcm[0];
if(nsamples % 1000 == 0) waveBuf[0].nsamples = ret;
fprintf(stderr, "\rRet: %d, Sample: %li", ret, (long)nsamples); ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
while(ndspChnIsPlaying(CHANNEL) == false)
{}
while(ndspChnIsPlaying(CHANNEL) == true)
{}
printf(" Buf0: %s.", waveBuf[0].status == NDSP_WBUF_QUEUED ? "Queued" : "Playing");
#endif
} }
gfxFlushBuffers();
gfxSwapBuffers();
gspWaitForVBlank();
free(pcm); free(pcm);
free(out);
if(ret == EXIT_SUCCESS) if(ret == EXIT_SUCCESS)
{ {
@@ -140,6 +168,8 @@ int convOpus(const char* in, const char *outf)
if(fclose(wav) != 0) if(fclose(wav) != 0)
fprintf(stderr, "\nError closing file: %d - %s\n", errno, strerror(errno)); fprintf(stderr, "\nError closing file: %d - %s\n", errno, strerror(errno));
out:
ndspExit();
op_free(of); op_free(of);
return ret; return ret;