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Refactor WAV support

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Only tested stereo WAV.

Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
Mahyar Koshkouei
2017-01-10 23:23:34 +00:00
parent d2b0ae540a
commit 06ad05e332
3 changed files with 129 additions and 174 deletions
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6
source/playback.c
View File

@@ -23,14 +23,12 @@ int playFile(const char* file)
switch(getFileType(file))
{
case FILE_TYPE_WAV:
playWav(file);
return 0;
setWav(&decoder);
break;
case FILE_TYPE_FLAC:
setFlac(&decoder);
break;
//playFlac(file);
//return 0;
case FILE_TYPE_OPUS:
setOpus(&decoder);

255
source/wav.c
View File

@@ -6,196 +6,113 @@
#include "all.h"
#include "wav.h"
#define BUFFER_SIZE (16 * 1024)
static const int buffSize = 16 * 1024;
static FILE* pWav = NULL;
static char header[45];
static uint8_t channels;
/**
* Plays a WAV file.
* Set decoder parameters for WAV.
*
* \param file File location of WAV file.
* \return Zero if successful, else failure.
* \param decoder Structure to store parameters.
*/
int playWav(const char *wav)
void setWav(struct decoder_fn* decoder)
{
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;
decoder->init = &initWav;
decoder->rate = &rateWav;
decoder->channels = &channelWav;
decoder->buffSize = buffSize;
decoder->decode = &readWav;
decoder->exit = &exitWav;
}
if(R_FAILED(ndspInit()))
{
err_print("Initialising ndsp failed.");
goto out;
}
/**
* Initialise WAV playback.
*
* \param file Location of WAV file to play.
* \return 0 on success, else failure.
*/
int initWav(const char* file)
{
pWav = fopen(file, "rb");
// TODO: Check if this is required.
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
if(file == NULL)
{
err_print("Opening file failed.");
goto out;
}
if(pWav == NULL)
return -1;
/* 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;
}
if(fread(header, 1, 44, pWav) == 0)
return -1;
/**
* 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]);
*/
format = (header[19]<<8) + (header[20]);
/* TODO: This should be moved to get file type */
/* Only support 16 bit PCM WAV */
if(((header[35]<<8) + (header[34])) != 16)
return -1;
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]);
#ifdef DEBUG
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);
#endif
if(channels > 2)
switch(channels)
{
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;
/* Only Mono and Stereo allowed */
case 1:
case 2:
break;
default:
printf("Bitness of %d unsupported.\n", bitness);
goto out;
return -1;
}
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;
printf("\33[2K\r%s", playing == false ? "Paused" : "");
}
if(playing == false || lastbuf == true)
continue;
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;
}
/**
* Get sampling rate of Wav file.
*
* \return Sampling rate.
*/
uint32_t rateWav(void)
{
return (header[27]<<24) + (header[26]<<16) + (header[25]<<8) +
(header[24]);
}
/**
* Get number of channels of Wav file.
*
* \return Number of channels for opened file.
*/
uint8_t channelWav(void)
{
return (header[23]<<8) + (header[22]);
}
/**
* Read part of open Wav file.
*
* \param buffer Output.
* \return Samples read for each channel.
*/
uint64_t readWav(void* buffer)
{
return fread(buffer, 1, buffSize, pWav) / channels;
}
/**
* Free Wav file.
*/
void exitWav(void)
{
fclose(pWav);
}

42
source/wav.h
View File

@@ -1 +1,41 @@
int playWav(const char *wav);
/**
* Set decoder parameters for WAV.
*
* \param decoder Structure to store parameters.
*/
void setWav(struct decoder_fn* decoder);
/**
* Initialise WAV playback.
*
* \param file Location of WAV file to play.
* \return 0 on success, else failure.
*/
int initWav(const char* file);
/**
* Get sampling rate of Wav file.
*
* \return Sampling rate.
*/
uint32_t rateWav(void);
/**
* Get number of channels of Wav file.
*
* \return Number of channels for opened file.
*/
uint8_t channelWav(void);
/**
* Read part of open Wav file.
*
* \param buffer Output.
* \return Samples read for each channel.
*/
uint64_t readWav(void* buffer);
/**
* Free Wav file.
*/
void exitWav(void);