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Homebrew Homebrew Development

  • Thread starter Thread starter aliak11
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Simple is a relative term. What platform are you targeting? There are so many takes on ogg decoders but without knowing what hardware you want to use it on there is no way to search for an easy solution for you.
 
PewnyPL said:
Don't quote me on this, but I'm fairly sure he wants to use it on, you know, 3DS. Since this is the 3DS homebrew development section.
Click to expand...

There have been OT talks of general programming info so I wanted the clarification. :P

MyLegGuy said:
Yeah, I was hoping for one already ported to the 3ds because I'm not experienced with audio enough to port one myself.

By simple, I mean something like this C code I made up that's similar to SoLoud.
Code: 
OggSound myGoodOggSound;
OggStream myGoodOggStream;

PlayHandle myPlayHandle1;
PlayHandle myPlayHandle2;

initAudio(OGG);

loadSound(&myGoodOggSound,"sfx.ogg");
loadStream(&myGoodOggStream,"music.ogg");

myPlayHandle1= playSound(myGoodOggSound);
myPlayHandle2 = playStream(myGoodOggStream);

while(1);

If there are no alternatives, I'll try SDL_Mixer. I don't want to though because it can only play one music stream at a time.
Click to expand...

LPP has ogg support that you could borrow from. Here is the piece of code that handles the stream decoding.
 
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MyLegGuy said:
Yeah, I was hoping for one already ported to the 3ds because I'm not experienced with audio enough to port one myself.

By simple, I mean something like this C code I made up that's similar to SoLoud.
Code: 
OggSound myGoodOggSound;
OggStream myGoodOggStream;

PlayHandle myPlayHandle1;
PlayHandle myPlayHandle2;

initAudio(OGG);

loadSound(&myGoodOggSound,"sfx.ogg");
loadStream(&myGoodOggStream,"music.ogg");

myPlayHandle1= playSound(myGoodOggSound);
myPlayHandle2 = playStream(myGoodOggStream);

while(1);

If there are no alternatives, I'll try SDL_Mixer. I don't want to though because it can only play one music stream at a time.
Click to expand...
You could also borrow some some of the code from ctrmus. It's licensed under the GPL v3.0.
 
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I attempted to play the ogg file using libtremor but failed.
Is there a mistake in the code?
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define MUSIC_CHANNEL 1
#define BUFFER_SIZE 4096
#define STACKSIZE (4 * 1024)

typedef struct {
	
	float rate;
	u32 channels;
	u32 encoding;
	u32 nsamples;
	u32 size;
	char* data;
	bool loop;
	int audiochannel;

	float mix[12];
	ndspInterpType interp;

	OggVorbis_File ovf;
} Music;
Music music;

void load() {
	FILE * file = fopen("example.ogg", "rb");
	if (ov_open(file, &music.ovf, NULL, 0) < 0) {
		printf("ogg vorbis file error\n");
	}
	vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
	if (vorbisInfo == NULL) {
		printf("could not retrieve ogg audio stream information\n");
	}
	music.rate = (float)vorbisInfo->rate;
	music.channels = (u32)vorbisInfo->channels;
	music.encoding = NDSP_ENCODING_PCM16;
	music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
	music.size = music.nsamples * music.channels * 2;
	music.audiochannel = 0;
	music.loop = false;
	if (linearSpaceFree() < music.size) {
		printf("not enough linear memory available\n");
	}
	music.data = (char*)linearAlloc(music.size);

	printf("rate:%f\n", music.rate);
	printf("channels:%ld\n", music.channels);
	printf("encoding:%ld\n", music.encoding);
	printf("nsamples:%ld\n", music.nsamples);
	printf("size:%ld\n", music.size);

	int offset = 0;
	int eof = 0;
	int currentSection;

	while (!eof) {
		long ret = ov_read(&music.ovf, &music.data[offset], 4096, &currentSection);
		if (ret == 0) {
			eof = 1;
		}
		else if (ret < 0) {
			ov_clear(&music.ovf);
			linearFree(music.data);
			printf("error in the ogg vorbis stream\n");
		}
		else {
			offset += ret;
		}
	}
	linearFree(&music.ovf);
	ov_clear(&music.ovf);
	fclose(file);
}

int play() {
	if (music.audiochannel == -1) {
		printf("No available audio channel\n");
		return -1;
	}
	ndspChnWaveBufClear(music.audiochannel);
	ndspChnReset(music.audiochannel);
	ndspChnInitParams(music.audiochannel);
	ndspChnSetMix(music.audiochannel, music.mix);
	ndspChnSetInterp(music.audiochannel, music.interp);
	ndspChnSetRate(music.audiochannel, music.rate);
	ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));
	
	ndspWaveBuf waveBuf;

	waveBuf.data_vaddr = music.data;
	waveBuf.nsamples = music.nsamples;
	waveBuf.looping = music.loop;

	DSP_FlushDataCache((u32*)music.data, music.size);

	ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
	return 0;
}


int main() {
	gfxInitDefault();
	consoleInit(GFX_TOP, nullptr);

	load();
	play();
	while (1) {
		hidScanInput();
		u32 kDown = hidKeysDown();
		if (kDown & KEY_START) break;

		gfxFlushBuffers();
		gfxSwapBuffers();
		gspWaitForVBlank();
	}
	gfxExit();
	ndspExit();
	return 0;
}
 
tgaiu said:
I attempted to play the ogg file using libtremor but failed.
Is there a mistake in the code?
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define MUSIC_CHANNEL 1
#define BUFFER_SIZE 4096
#define STACKSIZE (4 * 1024)

typedef struct {
   
    float rate;
    u32 channels;
    u32 encoding;
    u32 nsamples;
    u32 size;
    char* data;
    bool loop;
    int audiochannel;

    float mix[12];
    ndspInterpType interp;

    OggVorbis_File ovf;
} Music;
Music music;

void load() {
    FILE * file = fopen("example.ogg", "rb");
    if (ov_open(file, &music.ovf, NULL, 0) < 0) {
        printf("ogg vorbis file error\n");
    }
    vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
    if (vorbisInfo == NULL) {
        printf("could not retrieve ogg audio stream information\n");
    }
    music.rate = (float)vorbisInfo->rate;
    music.channels = (u32)vorbisInfo->channels;
    music.encoding = NDSP_ENCODING_PCM16;
    music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
    music.size = music.nsamples * music.channels * 2;
    music.audiochannel = 0;
    music.loop = false;
    if (linearSpaceFree() < music.size) {
        printf("not enough linear memory available\n");
    }
    music.data = (char*)linearAlloc(music.size);

    printf("rate:%f\n", music.rate);
    printf("channels:%ld\n", music.channels);
    printf("encoding:%ld\n", music.encoding);
    printf("nsamples:%ld\n", music.nsamples);
    printf("size:%ld\n", music.size);

    int offset = 0;
    int eof = 0;
    int currentSection;

    while (!eof) {
        long ret = ov_read(&music.ovf, &music.data[offset], 4096, &currentSection);
        if (ret == 0) {
            eof = 1;
        }
        else if (ret < 0) {
            ov_clear(&music.ovf);
            linearFree(music.data);
            printf("error in the ogg vorbis stream\n");
        }
        else {
            offset += ret;
        }
    }
    linearFree(&music.ovf);
    ov_clear(&music.ovf);
    fclose(file);
}

int play() {
    if (music.audiochannel == -1) {
        printf("No available audio channel\n");
        return -1;
    }
    ndspChnWaveBufClear(music.audiochannel);
    ndspChnReset(music.audiochannel);
    ndspChnInitParams(music.audiochannel);
    ndspChnSetMix(music.audiochannel, music.mix);
    ndspChnSetInterp(music.audiochannel, music.interp);
    ndspChnSetRate(music.audiochannel, music.rate);
    ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));
   
    ndspWaveBuf waveBuf;

    waveBuf.data_vaddr = music.data;
    waveBuf.nsamples = music.nsamples;
    waveBuf.looping = music.loop;

    DSP_FlushDataCache((u32*)music.data, music.size);

    ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
    return 0;
}


int main() {
    gfxInitDefault();
    consoleInit(GFX_TOP, nullptr);

    load();
    play();
    while (1) {
        hidScanInput();
        u32 kDown = hidKeysDown();
        if (kDown & KEY_START) break;

        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
    gfxExit();
    ndspExit();
    return 0;
}
Click to expand...
I don’t see a call to ndspInit?
 
elhobbs said:
I don’t see a call to ndspInit?
Click to expand...
I made those improvements but got errors.
Code: 
[HW.Memory <Error> core\memory.cpp:GetPointer:283: unknown GetPointer @ 0x00000000
[Audio.DSP <Warning> audio_core\hle\source.cpp:DequeueBuffer:312: source_id=0 buffer_id=20 length=0: Invalid physical address 0x00000000
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define MUSIC_CHANNEL 1
#define BUFFER_SIZE 4096
#define STACKSIZE (4 * 1024)

typedef struct {

	float rate;
	u32 channels;
	u32 encoding;
	u32 nsamples;
	u32 size;
	u8* data;
	bool loop;
	int audiochannel;

	float mix[12];
	ndspInterpType interp;

	OggVorbis_File ovf;
} Music;
Music music;

void load() {
	FILE * file = fopen("example.ogg", "rb");
	if (ov_open(file, &music.ovf, NULL, 0) < 0) {
		printf("ogg vorbis file error\n");
	}
	vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
	if (vorbisInfo == NULL) {
		printf("could not retrieve ogg audio stream information\n");
	}
	music.rate = (float)vorbisInfo->rate;
	music.channels = (u32)vorbisInfo->channels;
	music.encoding = NDSP_ENCODING_PCM16;
	music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
	music.size = music.nsamples * music.channels * 2;
	music.audiochannel = 0;
	music.loop = false;
	if (linearSpaceFree() < music.size) {
		printf("not enough linear memory available\n");
	music.data = static_cast<u8*>(linearAlloc(music.size));

	printf("rate:%f\n", music.rate);
	printf("channels:%ld\n", music.channels);
	printf("encoding:%ld\n", music.encoding);
	printf("nsamples:%ld\n", music.nsamples);
	printf("size:%ld\n", music.size);

	int offset = 0;
	int eof = 0;
	int currentSection;

	while (!eof) {
		long ret = ov_read(&music.ovf, &music.data[offset], 4096, &currentSection);
		if (ret == 0) {
			eof = 1;
		}
		else if (ret < 0) {
			ov_clear(&music.ovf);
			linearFree(music.data);
			printf("error in the ogg vorbis stream\n");
		}
		else {
			offset += ret;
		}
	}
	linearFree(&music.ovf);
	ov_clear(&music.ovf);
	fclose(file);
}

int play() {
	if (music.audiochannel == -1) {
		printf("No available audio channel\n");
		return -1;
	}
	ndspChnWaveBufClear(music.audiochannel);
	ndspChnReset(music.audiochannel);
	ndspChnInitParams(music.audiochannel);
	ndspChnSetMix(music.audiochannel, music.mix);
	ndspChnSetInterp(music.audiochannel, music.interp);
	ndspChnSetRate(music.audiochannel, music.rate);
	ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));

	ndspWaveBuf waveBuf;
	memset(&waveBuf, 0, sizeof(ndspWaveBuf));
	waveBuf.data_vaddr = music.data;
	waveBuf.nsamples = music.nsamples;
	waveBuf.looping = music.loop;
	waveBuf.status = NDSP_WBUF_FREE;

	DSP_FlushDataCache(music.data, music.size);

	ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
	return 0;
}


int main() {
	gfxInitDefault();
	consoleInit(GFX_TOP, nullptr);

	ndspInit();
	ndspSetOutputMode(NDSP_OUTPUT_STEREO);
	ndspSetOutputCount(1);

	load();
	play();
	while (aptMainLoop()) {
		hidScanInput();
		u32 keys = hidKeysDown();
		if (keys & KEY_START)
			break;

		gfxFlushBuffers();
		gfxSwapBuffers();
		gspWaitForVBlank();
	}
	ndspChnWaveBufClear(music.audiochannel);
	gfxExit();
	ndspExit();
	return 0;
}
 
tgaiu said:
I made those improvements but got errors.
Code: 
[HW.Memory <Error> core\memory.cpp:GetPointer:283: unknown GetPointer @ 0x00000000
[Audio.DSP <Warning> audio_core\hle\source.cpp:DequeueBuffer:312: source_id=0 buffer_id=20 length=0: Invalid physical address 0x00000000
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define MUSIC_CHANNEL 1
#define BUFFER_SIZE 4096
#define STACKSIZE (4 * 1024)

typedef struct {

    float rate;
    u32 channels;
    u32 encoding;
    u32 nsamples;
    u32 size;
    u8* data;
    bool loop;
    int audiochannel;

    float mix[12];
    ndspInterpType interp;

    OggVorbis_File ovf;
} Music;
Music music;

void load() {
    FILE * file = fopen("example.ogg", "rb");
    if (ov_open(file, &music.ovf, NULL, 0) < 0) {
        printf("ogg vorbis file error\n");
    }
    vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
    if (vorbisInfo == NULL) {
        printf("could not retrieve ogg audio stream information\n");
    }
    music.rate = (float)vorbisInfo->rate;
    music.channels = (u32)vorbisInfo->channels;
    music.encoding = NDSP_ENCODING_PCM16;
    music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
    music.size = music.nsamples * music.channels * 2;
    music.audiochannel = 0;
    music.loop = false;
    if (linearSpaceFree() < music.size) {
        printf("not enough linear memory available\n");
    music.data = static_cast<u8*>(linearAlloc(music.size));

    printf("rate:%f\n", music.rate);
    printf("channels:%ld\n", music.channels);
    printf("encoding:%ld\n", music.encoding);
    printf("nsamples:%ld\n", music.nsamples);
    printf("size:%ld\n", music.size);

    int offset = 0;
    int eof = 0;
    int currentSection;

    while (!eof) {
        long ret = ov_read(&music.ovf, &music.data[offset], 4096, &currentSection);
        if (ret == 0) {
            eof = 1;
        }
        else if (ret < 0) {
            ov_clear(&music.ovf);
            linearFree(music.data);
            printf("error in the ogg vorbis stream\n");
        }
        else {
            offset += ret;
        }
    }
    linearFree(&music.ovf);
    ov_clear(&music.ovf);
    fclose(file);
}

int play() {
    if (music.audiochannel == -1) {
        printf("No available audio channel\n");
        return -1;
    }
    ndspChnWaveBufClear(music.audiochannel);
    ndspChnReset(music.audiochannel);
    ndspChnInitParams(music.audiochannel);
    ndspChnSetMix(music.audiochannel, music.mix);
    ndspChnSetInterp(music.audiochannel, music.interp);
    ndspChnSetRate(music.audiochannel, music.rate);
    ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));

    ndspWaveBuf waveBuf;
    memset(&waveBuf, 0, sizeof(ndspWaveBuf));
    waveBuf.data_vaddr = music.data;
    waveBuf.nsamples = music.nsamples;
    waveBuf.looping = music.loop;
    waveBuf.status = NDSP_WBUF_FREE;

    DSP_FlushDataCache(music.data, music.size);

    ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
    return 0;
}


int main() {
    gfxInitDefault();
    consoleInit(GFX_TOP, nullptr);

    ndspInit();
    ndspSetOutputMode(NDSP_OUTPUT_STEREO);
    ndspSetOutputCount(1);

    load();
    play();
    while (aptMainLoop()) {
        hidScanInput();
        u32 keys = hidKeysDown();
        if (keys & KEY_START)
            break;

        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
    ndspChnWaveBufClear(music.audiochannel);
    gfxExit();
    ndspExit();
    return 0;
}
Click to expand...
Check if linearAlloc succeeded?
Why are you calling linearFree in the load function - on non “linear” region memory(music.ovf)?
 
elhobbs said:
Check if linearAlloc succeeded?
Why are you calling linearFree in the load function - on non “linear” region memory(music.ovf)?
Click to expand...
I made a mistake in placing linearFree.
linearAlloc seemed to be successful.
It seems that gspWaitForVBlank () is causing problems, but is it OK to delete this?
 
tgaiu said:
I made a mistake in placing linearFree.
linearAlloc seemed to be successful.
It seems that gspWaitForVBlank () is causing problems, but is it OK to delete this?
Click to expand...
If gspWaitForVBlank is having problems - then something else is causing it.
 
guys i know this sounds silly but i am new to the modding scene and also to gba temp . BUT I DONT KNOW HOW TO POST A FU#%ING question!!!!!!!!!!!!!!!HOW DO U POST ON GBATEMP NET!!!!!!!!!!!!!!!!!!
 
niBBasian said:
guys i know this sounds silly but i am new to the modding scene and also to gba temp . BUT I DONT KNOW HOW TO POST A FU#%ING question!!!!!!!!!!!!!!!HOW DO U POST ON GBATEMP NET!!!!!!!!!!!!!!!!!!
Click to expand...
aoyNMDx.png
 
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tgaiu said:
I made those improvements but got errors.
Code: 
[HW.Memory <Error> core\memory.cpp:GetPointer:283: unknown GetPointer @ 0x00000000
[Audio.DSP <Warning> audio_core\hle\source.cpp:DequeueBuffer:312: source_id=0 buffer_id=20 length=0: Invalid physical address 0x00000000
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define MUSIC_CHANNEL 1
#define BUFFER_SIZE 4096
#define STACKSIZE (4 * 1024)

typedef struct {

    float rate;
    u32 channels;
    u32 encoding;
    u32 nsamples;
    u32 size;
    u8* data;
    bool loop;
    int audiochannel;

    float mix[12];
    ndspInterpType interp;

    OggVorbis_File ovf;
} Music;
Music music;

void load() {
    FILE * file = fopen("example.ogg", "rb");
    if (ov_open(file, &music.ovf, NULL, 0) < 0) {
        printf("ogg vorbis file error\n");
    }
    vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
    if (vorbisInfo == NULL) {
        printf("could not retrieve ogg audio stream information\n");
    }
    music.rate = (float)vorbisInfo->rate;
    music.channels = (u32)vorbisInfo->channels;
    music.encoding = NDSP_ENCODING_PCM16;
    music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
    music.size = music.nsamples * music.channels * 2;
    music.audiochannel = 0;
    music.loop = false;
    if (linearSpaceFree() < music.size) {
        printf("not enough linear memory available\n");
    music.data = static_cast<u8*>(linearAlloc(music.size));

    printf("rate:%f\n", music.rate);
    printf("channels:%ld\n", music.channels);
    printf("encoding:%ld\n", music.encoding);
    printf("nsamples:%ld\n", music.nsamples);
    printf("size:%ld\n", music.size);

    int offset = 0;
    int eof = 0;
    int currentSection;

    while (!eof) {
        long ret = ov_read(&music.ovf, &music.data[offset], 4096, &currentSection);
        if (ret == 0) {
            eof = 1;
        }
        else if (ret < 0) {
            ov_clear(&music.ovf);
            linearFree(music.data);
            printf("error in the ogg vorbis stream\n");
        }
        else {
            offset += ret;
        }
    }
    linearFree(&music.ovf);
    ov_clear(&music.ovf);
    fclose(file);
}

int play() {
    if (music.audiochannel == -1) {
        printf("No available audio channel\n");
        return -1;
    }
    ndspChnWaveBufClear(music.audiochannel);
    ndspChnReset(music.audiochannel);
    ndspChnInitParams(music.audiochannel);
    ndspChnSetMix(music.audiochannel, music.mix);
    ndspChnSetInterp(music.audiochannel, music.interp);
    ndspChnSetRate(music.audiochannel, music.rate);
    ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));

    ndspWaveBuf waveBuf;
    memset(&waveBuf, 0, sizeof(ndspWaveBuf));
    waveBuf.data_vaddr = music.data;
    waveBuf.nsamples = music.nsamples;
    waveBuf.looping = music.loop;
    waveBuf.status = NDSP_WBUF_FREE;

    DSP_FlushDataCache(music.data, music.size);

    ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
    return 0;
}


int main() {
    gfxInitDefault();
    consoleInit(GFX_TOP, nullptr);

    ndspInit();
    ndspSetOutputMode(NDSP_OUTPUT_STEREO);
    ndspSetOutputCount(1);

    load();
    play();
    while (aptMainLoop()) {
        hidScanInput();
        u32 keys = hidKeysDown();
        if (keys & KEY_START)
            break;

        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
    ndspChnWaveBufClear(music.audiochannel);
    gfxExit();
    ndspExit();
    return 0;
}
Click to expand...

I tried a variety of hands, but I get a similar error.
Please let me know if you know the solution.
Code: 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

typedef struct {
	float rate;
	u32 channels;
	u32 encoding;
	u32 nsamples;
	u32 size;
	char* data;
	bool loop;
	int audiochannel;
	float mix[12];
	ndspInterpType interp;
	OggVorbis_File ovf;
} Music;
Music music;

void load() {
	FILE * file = fopen("test.ogg", "rb");
	if (ov_open(file, &music.ovf, NULL, 0) < 0) {
		printf("ogg vorbis file error\n");
	}
	vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
	if (vorbisInfo == NULL) {
		printf("could not retrieve ogg audio stream information\n");
	}
	music.rate = (float)vorbisInfo->rate;
	music.channels = (u32)vorbisInfo->channels;
	music.encoding = NDSP_ENCODING_PCM16;
	music.nsamples = (u32)ov_pcm_total(&music.ovf, -1);
	music.size = music.nsamples * music.channels * 2;
	music.audiochannel = 0;
	music.loop = false;
	music.interp = NDSP_INTERP_NONE;
	if (linearSpaceFree() < music.size) {
		printf("not enough linear memory available\n");
	}
	music.data = static_cast<char*>(linearAlloc(music.size));
	if (!music.data){
		fclose(file);
	}
	printf("buffersize:%d\n", linearGetSize(music.data));
	printf("rate:%f\n", music.rate);
	printf("channels:%ld\n", music.channels);
	printf("encoding:%ld\n", music.encoding);
	printf("nsamples:%ld\n", music.nsamples);
	printf("size:%ld\n", music.size);

	// Decoding loop
	int offset = 0;
	int eof = 0;
	int currentSection = 0;
	int readSize = 0;
	int comSize = 0;
	char* tmpBuffer = static_cast<char*>(linearAlloc(music.size));
	while (1) {
		readSize = ov_read(&music.ovf, (char*)tmpBuffer, 4096, &currentSection);
		if (readSize == 0) break; //EOF
		memcpy(music.data + comSize, tmpBuffer, readSize);
		comSize += readSize;
	}
	fclose(file);
}

int play() {
	if (music.audiochannel == -1) {
		printf("No available audio channel\n");
		return -1;
	}
	ndspChnReset(music.audiochannel);
	ndspChnInitParams(music.audiochannel);
	ndspChnSetMix(music.audiochannel, music.mix);
	ndspChnSetInterp(music.audiochannel, music.interp);
	ndspChnSetRate(music.audiochannel, music.rate);
	ndspChnSetFormat(music.audiochannel, NDSP_CHANNELS(music.channels) | NDSP_ENCODING(music.encoding));
	
	ndspWaveBuf waveBuf;
	memset(&waveBuf, 0, sizeof(ndspWaveBuf));
	waveBuf.data_vaddr = (const void*)music.data;
	waveBuf.nsamples = music.nsamples;
	waveBuf.looping = music.loop;
	waveBuf.status = NDSP_WBUF_FREE;
	DSP_FlushDataCache(music.data, music.size);
	ndspChnWaveBufAdd(music.audiochannel, &waveBuf);
	return 0;
}


int main() {
	gfxInitDefault();
	consoleInit(GFX_TOP, nullptr);

	ndspInit();
	ndspSetOutputMode(NDSP_OUTPUT_STEREO);
	ndspSetOutputCount(1);

	load();
	play();
	while (aptMainLoop()) {
		hidScanInput();
		u32 keys = hidKeysDown();
		if (keys & KEY_START)
			break;
		gfxFlushBuffers();
		gfxSwapBuffers();
		gspWaitForVBlank();
	}
	linearFree(&music.ovf);
	ov_clear(&music.ovf);
	ndspChnWaveBufClear(music.audiochannel);
	gfxExit();
	ndspExit();
	return 0;
}
 
That's not necessarily a problem with your code. Even legit games have that error in Citra. You will need to use the real hardware to test it.
 

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