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Refactored WAV code in to its own file

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Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
Mahyar Koshkouei
2016-12-29 21:15:19 +00:00
parent 42b61ba7f0
commit aa39c2f3fb
7 changed files with 222 additions and 208 deletions
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12
source/all.h Normal file
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@@ -0,0 +1,12 @@
#include <errno.h>
/* Channel to play music on */
#define CHANNEL 0x08
/* Adds extra debugging text */
#define DEBUG 0
/* Prints more error information */
#define err_print(err) \
do { fprintf(stderr, "\nError %d:%s(): %s %s\n", __LINE__, __func__, \
err, strerror(errno)); } while (0)

2
source/flac.c
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@@ -2,9 +2,9 @@
#define DR_FLAC_IMPLEMENTATION
#include <./dr_libs/dr_flac.h>
#include "all.h"
#define SAMPLES_TO_READ (16 * 1024)
#define CHANNEL 0x08
int playFlac(const char* in)
{

209
source/main.c
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@@ -9,27 +9,18 @@
#include <3ds.h>
#include <dirent.h>
#include <errno.h>
#include <opus/opusfile.h>
#include <stdbool.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "all.h"
#include "flac.h"
#include "main.h"
#include "opus.h"
#include "flac.h"
#include "wav.h"
#define BUFFER_SIZE (16 * 1024)
#define AUDIO_FOLDER "sdmc:/MUSIC/"
#define CHANNEL 0x08
/* Adds extra debugging text */
#define DEBUG 0
#define err_print(err) \
do { fprintf(stderr, "\nError %d:%s(): %s %s\n", __LINE__, __func__, \
err, strerror(errno)); } while (0)
#define AUDIO_FOLDER "sdmc:/MUSIC/"
enum file_types {
FILE_TYPE_ERROR = -1,
@@ -255,195 +246,3 @@ int getFileType(const char *file)
fclose(ftest);
return file_type;
}
/**
* Plays a WAV file.
*
* \param file File location of WAV file.
* \return Zero if successful, else failure.
*/
int playWav(const char *wav)
{
FILE* file = fopen(wav, "rb");
char header[45];
u32 sample;
u8 format;
u8 channels;
u8 bitness;
u32 byterate; // TODO: Not used.
u32 blockalign;
s16* buffer1 = NULL;
s16* buffer2 = NULL;
ndspWaveBuf waveBuf[2];
bool playing = true;
bool lastbuf = false;
if(R_FAILED(ndspInit()))
{
err_print("Initialising ndsp failed.");
goto out;
}
// TODO: Check if this is required.
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
if(file == NULL)
{
err_print("Opening file failed.");
goto out;
}
/* TODO: No need to read the first number of bytes */
if(fread(header, 1, 44, file) == 0)
{
err_print("Unable to read WAV file.");
goto out;
}
/**
* http://www.topherlee.com/software/pcm-tut-wavformat.html and
* http://soundfile.sapp.org/doc/WaveFormat/ helped a lot.
*/
format = (header[19]<<8) + (header[20]);
channels = (header[23]<<8) + (header[22]);
sample = (header[27]<<24) + (header[26]<<16) + (header[25]<<8) +
(header[24]);
byterate = (header[31]<<24) + (header[30]<<16) + (header[29]<<8) +
(header[28]);
blockalign = (header[33]<<8) + (header[32]);
bitness = (header[35]<<8) + (header[34]);
printf("Format: %s(%d), Ch: %d, Sam: %lu, bit: %d, BR: %lu, BA: %lu\n",
format == 1 ? "PCM" : "Other", format, channels, sample, bitness,
byterate, blockalign);
if(channels > 2)
{
puts("Error: Invalid number of channels.");
goto out;
}
/**
* Playing ADPCM, and 8 bit WAV files are disabled as they both sound like
* complete garbage.
*/
switch(bitness)
{
case 8:
bitness = channels == 2 ? NDSP_FORMAT_STEREO_PCM8 :
NDSP_FORMAT_MONO_PCM8;
puts("8bit playback disabled.");
goto out;
case 16:
bitness = channels == 2 ? NDSP_FORMAT_STEREO_PCM16 :
NDSP_FORMAT_MONO_PCM16;
break;
default:
printf("Bitness of %d unsupported.\n", bitness);
goto out;
}
ndspChnReset(CHANNEL);
ndspChnWaveBufClear(CHANNEL);
/* Polyphase sounds much better than linear or no interpolation */
ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
ndspChnSetRate(CHANNEL, sample);
ndspChnSetFormat(CHANNEL, bitness);
memset(waveBuf, 0, sizeof(waveBuf));
buffer1 = (s16*) linearAlloc(BUFFER_SIZE);
buffer2 = (s16*) linearAlloc(BUFFER_SIZE);
fread(buffer1, 1, BUFFER_SIZE, file);
waveBuf[0].nsamples = BUFFER_SIZE / blockalign;
waveBuf[0].data_vaddr = &buffer1[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
fread(buffer2, 1, BUFFER_SIZE, file);
waveBuf[1].nsamples = BUFFER_SIZE / blockalign;
waveBuf[1].data_vaddr = &buffer2[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
printf("Playing %s\n", wav);
/**
* There may be a chance that the music has not started by the time we get
* to the while loop. So we ensure that music has started here.
*/
while(ndspChnIsPlaying(CHANNEL) == false);
while(playing == false || ndspChnIsPlaying(CHANNEL) == true)
{
u32 kDown;
/* Number of bytes read from file.
* Static only for the purposes of the printf debug at the bottom.
*/
static size_t read = 0;
gfxSwapBuffers();
gfxFlushBuffers();
gspWaitForVBlank();
hidScanInput();
kDown = hidKeysDown();
if(kDown & KEY_B)
break;
if(kDown & (KEY_A | KEY_R))
playing = !playing;
if(playing == false || lastbuf == true)
{
printf("\33[2K\rPaused");
continue;
}
printf("\33[2K\r");
if(waveBuf[0].status == NDSP_WBUF_DONE)
{
read = fread(buffer1, 1, BUFFER_SIZE, file);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < BUFFER_SIZE)
waveBuf[0].nsamples = read / blockalign;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
}
if(waveBuf[1].status == NDSP_WBUF_DONE)
{
read = fread(buffer2, 1, BUFFER_SIZE, file);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < BUFFER_SIZE)
waveBuf[1].nsamples = read / blockalign;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
}
DSP_FlushDataCache(buffer1, BUFFER_SIZE);
DSP_FlushDataCache(buffer2, BUFFER_SIZE);
}
ndspChnWaveBufClear(CHANNEL);
out:
puts("Stopping playback.");
ndspExit();
fclose(file);
linearFree(buffer1);
linearFree(buffer2);
return 0;
}

3
source/opus.c
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@@ -1,12 +1,11 @@
#include <3ds.h>
#include <opus/opusfile.h>
#include <stdlib.h>
#include <string.h>
#include "all.h"
#include "opus.h"
#define SAMPLES_TO_READ (32 * 1024)
#define CHANNEL 0x08
int playOpus(const char* in)
{

2
source/opus.h
View File

@@ -1,3 +1,5 @@
#include <opus/opusfile.h>
int playOpus(const char* in);
uint64_t fillOpusBuffer(OggOpusFile* opusFile, uint64_t samplesToRead,

201
source/wav.c Normal file
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@@ -0,0 +1,201 @@
#include <3ds.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "all.h"
#include "wav.h"
#define BUFFER_SIZE (16 * 1024)
/**
* Plays a WAV file.
*
* \param file File location of WAV file.
* \return Zero if successful, else failure.
*/
int playWav(const char *wav)
{
FILE* file = fopen(wav, "rb");
char header[45];
u32 sample;
u8 format;
u8 channels;
u8 bitness;
u32 byterate; // TODO: Not used.
u32 blockalign;
s16* buffer1 = NULL;
s16* buffer2 = NULL;
ndspWaveBuf waveBuf[2];
bool playing = true;
bool lastbuf = false;
if(R_FAILED(ndspInit()))
{
err_print("Initialising ndsp failed.");
goto out;
}
// TODO: Check if this is required.
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
if(file == NULL)
{
err_print("Opening file failed.");
goto out;
}
/* TODO: No need to read the first number of bytes */
if(fread(header, 1, 44, file) == 0)
{
err_print("Unable to read WAV file.");
goto out;
}
/**
* http://www.topherlee.com/software/pcm-tut-wavformat.html and
* http://soundfile.sapp.org/doc/WaveFormat/ helped a lot.
*/
format = (header[19]<<8) + (header[20]);
channels = (header[23]<<8) + (header[22]);
sample = (header[27]<<24) + (header[26]<<16) + (header[25]<<8) +
(header[24]);
byterate = (header[31]<<24) + (header[30]<<16) + (header[29]<<8) +
(header[28]);
blockalign = (header[33]<<8) + (header[32]);
bitness = (header[35]<<8) + (header[34]);
printf("Format: %s(%d), Ch: %d, Sam: %lu, bit: %d, BR: %lu, BA: %lu\n",
format == 1 ? "PCM" : "Other", format, channels, sample, bitness,
byterate, blockalign);
if(channels > 2)
{
puts("Error: Invalid number of channels.");
goto out;
}
/**
* Playing ADPCM, and 8 bit WAV files are disabled as they both sound like
* complete garbage.
*/
switch(bitness)
{
case 8:
bitness = channels == 2 ? NDSP_FORMAT_STEREO_PCM8 :
NDSP_FORMAT_MONO_PCM8;
puts("8bit playback disabled.");
goto out;
case 16:
bitness = channels == 2 ? NDSP_FORMAT_STEREO_PCM16 :
NDSP_FORMAT_MONO_PCM16;
break;
default:
printf("Bitness of %d unsupported.\n", bitness);
goto out;
}
ndspChnReset(CHANNEL);
ndspChnWaveBufClear(CHANNEL);
/* Polyphase sounds much better than linear or no interpolation */
ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
ndspChnSetRate(CHANNEL, sample);
ndspChnSetFormat(CHANNEL, bitness);
memset(waveBuf, 0, sizeof(waveBuf));
buffer1 = (s16*) linearAlloc(BUFFER_SIZE);
buffer2 = (s16*) linearAlloc(BUFFER_SIZE);
fread(buffer1, 1, BUFFER_SIZE, file);
waveBuf[0].nsamples = BUFFER_SIZE / blockalign;
waveBuf[0].data_vaddr = &buffer1[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
fread(buffer2, 1, BUFFER_SIZE, file);
waveBuf[1].nsamples = BUFFER_SIZE / blockalign;
waveBuf[1].data_vaddr = &buffer2[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
printf("Playing %s\n", wav);
/**
* There may be a chance that the music has not started by the time we get
* to the while loop. So we ensure that music has started here.
*/
while(ndspChnIsPlaying(CHANNEL) == false);
while(playing == false || ndspChnIsPlaying(CHANNEL) == true)
{
u32 kDown;
/* Number of bytes read from file.
* Static only for the purposes of the printf debug at the bottom.
*/
static size_t read = 0;
gfxSwapBuffers();
gfxFlushBuffers();
gspWaitForVBlank();
hidScanInput();
kDown = hidKeysDown();
if(kDown & KEY_B)
break;
if(kDown & (KEY_A | KEY_R))
playing = !playing;
if(playing == false || lastbuf == true)
{
printf("\33[2K\rPaused");
continue;
}
printf("\33[2K\r");
if(waveBuf[0].status == NDSP_WBUF_DONE)
{
read = fread(buffer1, 1, BUFFER_SIZE, file);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < BUFFER_SIZE)
waveBuf[0].nsamples = read / blockalign;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
}
if(waveBuf[1].status == NDSP_WBUF_DONE)
{
read = fread(buffer2, 1, BUFFER_SIZE, file);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < BUFFER_SIZE)
waveBuf[1].nsamples = read / blockalign;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
}
DSP_FlushDataCache(buffer1, BUFFER_SIZE);
DSP_FlushDataCache(buffer2, BUFFER_SIZE);
}
ndspChnWaveBufClear(CHANNEL);
out:
puts("Stopping playback.");
ndspExit();
fclose(file);
linearFree(buffer1);
linearFree(buffer2);
return 0;
}

1
source/wav.h Normal file
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@@ -0,0 +1 @@
int playWav(const char *wav);