Homebrew Homebrew Development

tgaiu · Dec 15, 2017

MyLegGuy said:
Just a random idea, make sure you're loading a small OGG file because you're loading it all into memory at once.

Are you talking about streaming play?
Should I try it?

Deleted User · Dec 15, 2017

elhobbs · Dec 15, 2017

MyLegGuy said:
Just a random idea, make sure you're loading a small OGG file because you're loading it all into memory at once.

Also make sure you run the dsp dump utility on hardware. You will need to copy the file from hardware to the sd card folder for Citra - make sure you use the same path.

elhobbs · Dec 16, 2017

tgaiu said:
I tried it on 3DS but it ended with an error "an exception occurred".

here is some code that will work with small ogg files. the main issue was that waveBuf was going out of scope - it needs to be valid the entire time it is being used. also, you need to set the channel volume or you wont hear anything.

Code:

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#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;
ndspWaveBuf waveBuf;
void load() {
 memset(&music, 0, sizeof(music));
 music.mix[0] = 1.0f;
 music.mix[1] = 1.0f;
 FILE * file = fopen("example.ogg", "rb");
 if (file == 0) {
  printf("no file\n");
  while (1);
 }
 if (ov_open(file, &music.ovf, NULL, 0) < 0) {
  printf("ogg vorbis file error\n");
  while (1);
 }
 vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
 if (vorbisInfo == NULL) {
  printf("could not retrieve ogg audio stream information\n");
  while (1);
 }
 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.interp = NDSP_INTERP_NONE;
 music.loop = false;
 if (linearSpaceFree() < music.size) {
  printf("not enough linear memory available %ld\n", music.size);
 }
 music.data = (u8 *)linearAlloc(music.size);
 if (music.data == 0) {
  printf("null\n");
  while (1);
 }
 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");
   while (1);
  }
  else {
   offset += ret;
  }
  //printf("%ld %d\n", ret, currentSection);
 }
 printf("done\n");
 //linearFree(&music.ovf);
 ov_clear(&music.ovf);
 fclose(file);
}
int play() {
 if (music.audiochannel == -1) {
  printf("No available audio channel\n");
  return -1;
 }
 printf("music: %p\n,", music.data);
 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));
 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);
// while (1);
 return 0;
}

int main(int argc, char **argv) {
 gfxInitDefault();
 consoleInit(GFX_TOP, 0);
 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);
 ndspExit();
 gfxExit();
 return 0;
}

tgaiu · Dec 17, 2017

elhobbs said:

here is some code that will work with small ogg files. the main issue was that waveBuf was going out of scope - it needs to be valid the entire time it is being used. also, you need to set the channel volume or you wont hear anything.

Code:

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#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;
ndspWaveBuf waveBuf;
void load() {
 memset(&music, 0, sizeof(music));
 music.mix[0] = 1.0f;
 music.mix[1] = 1.0f;
 FILE * file = fopen("example.ogg", "rb");
 if (file == 0) {
  printf("no file\n");
  while (1);
 }
 if (ov_open(file, &music.ovf, NULL, 0) < 0) {
  printf("ogg vorbis file error\n");
  while (1);
 }
 vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
 if (vorbisInfo == NULL) {
  printf("could not retrieve ogg audio stream information\n");
  while (1);
 }
 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.interp = NDSP_INTERP_NONE;
 music.loop = false;
 if (linearSpaceFree() < music.size) {
  printf("not enough linear memory available %ld\n", music.size);
 }
 music.data = (u8 *)linearAlloc(music.size);
 if (music.data == 0) {
  printf("null\n");
  while (1);
 }
 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");
   while (1);
  }
  else {
   offset += ret;
  }
  //printf("%ld %d\n", ret, currentSection);
 }
 printf("done\n");
 //linearFree(&music.ovf);
 ov_clear(&music.ovf);
 fclose(file);
}
int play() {
 if (music.audiochannel == -1) {
  printf("No available audio channel\n");
  return -1;
 }
 printf("music: %p\n,", music.data);
 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));
 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);
// while (1);
 return 0;
}

int main(int argc, char **argv) {
 gfxInitDefault();
 consoleInit(GFX_TOP, 0);
 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);
 ndspExit();
 gfxExit();
 return 0;
}

I'm really thankful to you! I will refer to your code.

Deleted User · Dec 18, 2017

jockep · Dec 18, 2017

MyLegGuy said:
Can I just call exit(0) to exit my CIA homebrew, or do I actually have to clean up everything? (sdmcExit(), fsExit(), romfsExit(), etc.)

Pretty sure not doing that will leave resources in memory which is bad.

Togetoge · Dec 21, 2017

I played music files using ndsp but playback finished faster than playing on Windows.
Is there a way to prevent this?
Sorry, this problem has already been solved.

elhobbs · Dec 22, 2017

jockep said:
Pretty sure not doing that will leave resources in memory which is bad.

You need to exit everything you initialize yourself.

Deleted User · Dec 22, 2017

elhobbs · Dec 22, 2017

MyLegGuy said:
Do I also need to free any memory allocated with malloc, or can the 3ds free it itself after the program is closed?

No, the system releases the entire memory regions used by malloc and linearAlloc back to the system when your program ends.

Joel16 · Dec 22, 2017

elhobbs said:
No, the system releases the entire memory regions used by malloc and linearAlloc back to the system when your program ends.

True, although it's good practice to free once it's no longer being used.

MaiconErick · Dec 23, 2017

How can I search for all instaled titles and store their title IDs in a list?
I'm playing and learning with the DevKit's examples, but didn't find anything related.
Help

elhobbs · Dec 24, 2017

MaiconErick said:
How can I search for all instaled titles and store their title IDs in a list?
I'm playing and learning with the DevKit's examples, but didn't find anything related.
Help

You could look at the source for FBI and see how it does it.

MaiconErick · Dec 24, 2017

elhobbs said:
You could look at the source for FBI and see how it does it.

That's what I'm doing, I'm starting with 3DSident's "Installed title count", to have an idea of how it works first.
It's really the hard way, I don't know what they're doing, I'm just trying to follow along haha

I can't even find where the method(s) for listing the titles are in fbi source.

elhobbs · Dec 24, 2017

MaiconErick said:
That's what I'm doing, I'm starting with 3DSident's "Installed title count", to have an idea of how it works first.
It's really the hard way, I don't know what they're doing, I'm just trying to follow along haha

I can't even find where the method(s) for listing the titles are in fbi source.

https://github.com/Steveice10/FBI/blob/master/source/ui/section/task/listtitles.c

MaiconErick · Dec 24, 2017

I got my TitleID value stored in a u64 titleID[1] in decimal.
I need help on how to use it on this function:
APT_PrepareToDoApplicationJump(0, 0x000XLL, 0),
where X stands for the titleID value in hexadecimal.
Spent some hours already trying to look up how to do this using C, so I decided to ask for help here.

the titleID is 16 (decimal) numbers long.
the X value I need, should be 13 (hex) numbers long.

EDIT: Figured it out.

roytam1 · Jan 6, 2018

Is it possible to change receive buffer size when using httpc service to download file?

tgaiu · Jan 8, 2018

I attempted to play the stream of ogg files, but the playback speed got a little faster.
Please help me.

Code:

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <tremor/ivorbiscodec.h>
#include <tremor/ivorbisfile.h>
#include <3ds.h>

#define BUFF_SIZE 8 * 4096
typedef struct {
    float rate;
    u32 channels;
    u32 encoding;
    u32 nsamples;
    u32 size;
    char* data1;
    char* data2;
    bool loop;
    int audiochannel;
    float mix[12];
    ndspInterpType interp;
    OggVorbis_File ovf;
} Music;
Music music;
ndspWaveBuf waveBuf[2];

uint64_t fillVorbisBuffer(OggVorbis_File ovf, char* buffer) {
    uint64_t samplesRead = 0;
    int samplesToRead = BUFF_SIZE;

    while (samplesToRead > 0) {

        static int current_section;
        int samplesJustRead =
            ov_read(&ovf,
                buffer,
                samplesToRead > 4096 ? 4096 : samplesToRead,
                &current_section);
        if (samplesJustRead < 0)
            return samplesJustRead;
        else if (samplesJustRead == 0){
            break;
        }
        samplesRead += samplesJustRead;
        samplesToRead -= samplesJustRead;
        buffer += samplesJustRead;
    }
    return samplesRead / sizeof(int16_t);
}


int load() {
    memset(&music, 0, sizeof(music));
    music.mix[0] = 1.0f;
    music.mix[1] = 1.0f;
    FILE * file = fopen("example.ogg", "rb");
    if (file == 0) {
        printf("no file\n");
        while (1);
    }
    if (ov_open(file, &music.ovf, NULL, 0) < 0) {
        printf("ogg vorbis file error\n");
        while (1);
    }
    vorbis_info * vorbisInfo = ov_info(&music.ovf, -1);
    if (vorbisInfo == NULL) {
        printf("could not retrieve ogg audio stream information\n");
        while (1);
    }
    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.interp = NDSP_INTERP_NONE;
    music.loop = false;
    if (linearSpaceFree() < music.size) {
        printf("not enough linear memory available %ld\n", music.size);
    }
    music.data1 = (char *)linearAlloc(BUFF_SIZE * sizeof(int16_t));
    music.data2 = (char *)linearAlloc(BUFF_SIZE * sizeof(int16_t));
    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);
    //fclose(file);
    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));
    memset(&waveBuf, 0, sizeof(ndspWaveBuf));
    waveBuf[0].data_vaddr = &music.data1[0];
    waveBuf[0].nsamples = fillVorbisBuffer(music.ovf, &music.data1[0]) / music.channels;
    waveBuf[0].looping = music.loop;
    waveBuf[1].data_vaddr = &music.data2[0];
    waveBuf[1].nsamples = fillVorbisBuffer(music.ovf, &music.data1[0]) / music.channels;
    waveBuf[1].looping = music.loop;
    return 0;
}

int main(int argc, char **argv) {
    gfxInitDefault();
    consoleInit(GFX_TOP, 0);
    ndspInit();
    ndspSetOutputMode(NDSP_OUTPUT_STEREO);
    ndspSetOutputCount(1);
    load();
  
    ndspChnWaveBufAdd(0, &waveBuf[0]);
    ndspChnWaveBufAdd(0, &waveBuf[1]);

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

        if (waveBuf[0].status == NDSP_WBUF_DONE){
            size_t read = fillVorbisBuffer(music.ovf, &music.data1[0]);

            if (read <= 0){
                continue;
            }
            else if (read < BUFF_SIZE)
                waveBuf[0].nsamples = read / music.channels;

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

        if (waveBuf[1].status == NDSP_WBUF_DONE){
            size_t read = fillVorbisBuffer(music.ovf, &music.data2[0]);

            if (read <= 0){
                continue;
            }
            else if (read < BUFF_SIZE)
                waveBuf[1].nsamples = read / music.channels;

            ndspChnWaveBufAdd(music.audiochannel, &waveBuf[1]);
        }
        DSP_FlushDataCache(music.data1, BUFF_SIZE * sizeof(int16_t));
        DSP_FlushDataCache(music.data2, BUFF_SIZE * sizeof(int16_t));
        //printf("%d:%d\n", waveBuf[0].status, waveBuf[1].status);
        gfxFlushBuffers();
        gfxSwapBuffers();
        gspWaitForVBlank();
    }
    ndspChnWaveBufClear(music.audiochannel);
    ndspExit();
    gfxExit();
    return 0;
}

InvincibleLight · Jan 24, 2018

On a scale of 1 - 10, how difficult would it be to make Android for 3DS that could start on boot?

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Ils ne passeront pas

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