Refactor Opus decoder

Created generic playback handler, first with Opus support. Other
decoders to follow. This is to remove duplicated code.

3DSX tested working with citra and N3DS.

Signed-off-by: Mahyar Koshkouei <deltabeard@users.noreply.github.com>
This commit is contained in:
Mahyar Koshkouei
2017-01-07 23:26:34 +00:00
parent c5115935d9
commit 644e501d6e
7 changed files with 251 additions and 151 deletions

View File

@@ -4,9 +4,19 @@
#define CHANNEL 0x08
/* Adds extra debugging text */
//#define DEBUG
#define DEBUG
/* Prints more error information */
#define err_print(err) \
do { fprintf(stderr, "\nError %d:%s(): %s %s\n", __LINE__, __func__, \
err, strerror(errno)); } while (0)
struct decoder_fn
{
int (* init)(const char* file);
uint32_t (* rate)(void);
uint8_t channels;
int buffSize;
uint64_t (* decode)(void*);
void (* exit)(void);
};

View File

@@ -20,22 +20,9 @@
#include "main.h"
#include "mp3.h"
#include "opus.h"
#include "playback.h"
#include "wav.h"
/* Default folder */
#define DEFAULT_DIR "sdmc:/"
/* Maximum number of lines that can be displayed */
#define MAX_LIST 27
enum file_types {
FILE_TYPE_ERROR = -1,
FILE_TYPE_WAV,
FILE_TYPE_FLAC,
FILE_TYPE_OGG,
FILE_TYPE_OPUS,
FILE_TYPE_MP3
};
int main(int argc, char **argv)
{
PrintConsole topScreen;
@@ -204,6 +191,7 @@ int main(int argc, char **argv)
else
{
consoleSelect(&topScreen);
/* Move this to playback.c */
switch(getFileType(file))
{
case FILE_TYPE_WAV:
@@ -215,7 +203,7 @@ int main(int argc, char **argv)
break;
case FILE_TYPE_OPUS:
playOpus(file);
playFile(file);
break;
case FILE_TYPE_MP3:

View File

@@ -7,6 +7,21 @@
* LICENSE file.
*/
/* Default folder */
#define DEFAULT_DIR "sdmc:/"
/* Maximum number of lines that can be displayed */
#define MAX_LIST 27
enum file_types {
FILE_TYPE_ERROR = -1,
FILE_TYPE_WAV,
FILE_TYPE_FLAC,
FILE_TYPE_OGG,
FILE_TYPE_OPUS,
FILE_TYPE_MP3
};
/**
* Get number of files in current working folder
*

View File

@@ -7,146 +7,84 @@
#define SAMPLES_TO_READ (32 * 1024)
int playOpus(const char* in)
static OggOpusFile* opusFile;
static const OpusHead* opusHead;
/**
* Set decoder parameters for Opus.
*
* \param decoder Structure to store parameters.
*/
void setOpus(struct decoder_fn* decoder)
{
int err = 0;
int16_t* buffer1 = linearAlloc(SAMPLES_TO_READ * sizeof(int16_t));
int16_t* buffer2 = linearAlloc(SAMPLES_TO_READ * sizeof(int16_t));
ndspWaveBuf waveBuf[2];
bool playing = true;
bool lastbuf = false;
OggOpusFile* opusFile = op_open_file(in, &err);
const OpusHead* opusHead;
int link;
if(err != 0)
{
printf("libopusfile failed with error %d.", err);
return -1;
}
if(R_FAILED(ndspInit()))
{
printf("Initialising ndsp failed.");
goto out;
}
if((link = op_current_link(opusFile)) < 0)
{
printf("Error getting current link: %d\n", link);
goto out;
}
opusHead = op_head(opusFile, link);
#ifdef DEBUG
printf("\nRate: %lu\tChan: %d\n", opusHead->input_sample_rate,
opusHead->channel_count);
#endif
ndspChnReset(CHANNEL);
ndspChnWaveBufClear(CHANNEL);
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
ndspChnSetRate(CHANNEL, opusHead->input_sample_rate);
ndspChnSetFormat(CHANNEL, NDSP_FORMAT_STEREO_PCM16);
memset(waveBuf, 0, sizeof(waveBuf));
waveBuf[0].nsamples =
fillOpusBuffer(opusFile, SAMPLES_TO_READ, buffer1) / 2;
waveBuf[0].data_vaddr = &buffer1[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
waveBuf[1].nsamples =
fillOpusBuffer(opusFile, SAMPLES_TO_READ, buffer2) / 2;
waveBuf[1].data_vaddr = &buffer2[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
printf("Playing %s\n", in);
/**
* 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 = fillOpusBuffer(opusFile, SAMPLES_TO_READ, buffer1);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < SAMPLES_TO_READ)
waveBuf[0].nsamples = read / opusHead->channel_count;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
}
if(waveBuf[1].status == NDSP_WBUF_DONE)
{
read = fillOpusBuffer(opusFile, SAMPLES_TO_READ, buffer2);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < SAMPLES_TO_READ)
waveBuf[1].nsamples = read / opusHead->channel_count;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
}
DSP_FlushDataCache(buffer1, SAMPLES_TO_READ * sizeof(s16));
DSP_FlushDataCache(buffer2, SAMPLES_TO_READ * sizeof(s16));
}
out:
printf("\nStopping Opus playback.\n");
ndspChnWaveBufClear(CHANNEL);
ndspExit();
linearFree(buffer1);
linearFree(buffer2);
op_free(opusFile);
return 0;
decoder->init = initOpus;
decoder->rate = rateOpus;
/* Opus decoder always returns stereo stream */
decoder->channels = 2;
decoder->buffSize = SAMPLES_TO_READ;
decoder->decode = decodeOpus;
decoder->exit = exitOpus;
}
/**
* Fill a buffer with decoded samples.
* Initialise Opus decoder.
*
* \param opusFile OggOpusFile pointer.
* \param samplesToRead Number of samples to read in to buffer.
* \param bufferOut Pointer to output buffer.
* \return Number of samples read per channel.
* \param file Location of opus file to play.
* \return 0 on success, else failure.
*/
int initOpus(const char* file)
{
int err = 0;
if((opusFile = op_open_file(file, &err)) == NULL)
goto out;
if((err = op_current_link(opusFile)) < 0)
goto out;
opusHead = op_head(opusFile, err);
out:
return err;
}
/**
* Get sampling rate of Opus file.
*
* \return Sampling rate. Should be 48000.
*/
uint32_t rateOpus(void)
{
return opusHead->input_sample_rate;
}
/**
* Decode part of open Opus file.
*
* \param buffer Decoded output.
* \return Samples read for each channel.
*/
uint64_t decodeOpus(void* buffer)
{
return fillOpusBuffer(opusFile, SAMPLES_TO_READ, buffer);
}
/**
* Free Opus decoder.
*/
void exitOpus(void)
{
op_free(opusFile);
}
/**
* Decode Opus file to fill buffer.
*
* \param opusFile File to decode.
* \param samplesToRead Number of samples to read in to buffer. Must not exceed
* size of buffer.
* \param bufferOut Pointer to buffer.
* \return Samples read per channel.
*/
uint64_t fillOpusBuffer(OggOpusFile* opusFile, uint64_t samplesToRead,
int16_t* bufferOut)

View File

@@ -1,5 +1,15 @@
#include <opus/opusfile.h>
void setOpus(struct decoder_fn* decoder);
int initOpus(const char* file);
uint32_t rateOpus(void);
uint64_t decodeOpus(void* buffer);
void exitOpus(void);
int playOpus(const char* in);
uint64_t fillOpusBuffer(OggOpusFile* opusFile, uint64_t samplesToRead,

138
source/playback.c Normal file
View File

@@ -0,0 +1,138 @@
#include <3ds.h>
#include <stdlib.h>
#include <string.h>
#include "all.h"
#include "opus.h"
#include "playback.h"
int playFile(const char* file)
{
struct decoder_fn decoder;
int16_t* buffer1 = NULL;
int16_t* buffer2 = NULL;
ndspWaveBuf waveBuf[2];
bool playing = true;
bool lastbuf = false;
printf("Here: %d\n", __LINE__);
if(R_FAILED(ndspInit()))
{
printf("Initialising ndsp failed.");
goto out;
}
printf("Here: %d\n", __LINE__);
setOpus(&decoder);
printf("Here: %d\n", __LINE__);
buffer1 = linearAlloc(decoder.buffSize * sizeof(int16_t));
buffer2 = linearAlloc(decoder.buffSize * sizeof(int16_t));
printf("Here: %d\n", __LINE__);
if((*decoder.init)(file) != 0)
goto out;
printf("Here: %d\n", __LINE__);
#ifdef DEBUG
printf("\nRate: %lu\tChan: %d\n", (*decoder.rate)(), decoder.channels);
#endif
ndspChnReset(CHANNEL);
ndspChnWaveBufClear(CHANNEL);
ndspSetOutputMode(NDSP_OUTPUT_STEREO);
ndspChnSetInterp(CHANNEL, NDSP_INTERP_POLYPHASE);
ndspChnSetRate(CHANNEL, (*decoder.rate)());
ndspChnSetFormat(CHANNEL, NDSP_FORMAT_STEREO_PCM16);
printf("Here: %d\n", __LINE__);
memset(waveBuf, 0, sizeof(waveBuf));
waveBuf[0].nsamples = (*decoder.decode)(&buffer1[0]) / decoder.channels;
waveBuf[0].data_vaddr = &buffer1[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
printf("Here: %d\n", __LINE__);
waveBuf[1].nsamples = (*decoder.decode)(&buffer2[0]) / decoder.channels;
waveBuf[1].data_vaddr = &buffer2[0];
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
printf("Playing %s\n", file);
/**
* 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 = (*decoder.decode)(&buffer1[0]);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < decoder.buffSize)
waveBuf[0].nsamples = read / decoder.channels;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[0]);
}
if(waveBuf[1].status == NDSP_WBUF_DONE)
{
read = (*decoder.decode)(&buffer2[0]);
if(read == 0)
{
lastbuf = true;
continue;
}
else if(read < decoder.buffSize)
waveBuf[1].nsamples = read / decoder.channels;
ndspChnWaveBufAdd(CHANNEL, &waveBuf[1]);
}
DSP_FlushDataCache(buffer1, decoder.buffSize * sizeof(int16_t));
DSP_FlushDataCache(buffer2, decoder.buffSize * sizeof(int16_t));
}
out:
printf("\nStopping playback.\n");
(*decoder.exit)();
ndspChnWaveBufClear(CHANNEL);
ndspExit();
linearFree(buffer1);
linearFree(buffer2);
return 0;
}

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