ref: e218b68df7d26a9394f27252ff34c1d96c38e35f
dir: /src/Backends/Audio/SoftwareMixer/WiiU.cpp/
// Released under the MIT licence.
// See LICENCE.txt for details.
#include "Backend.h"
#include <math.h>
#include <stddef.h>
#include <stdlib.h>
#include <string.h>
#include <coreinit/cache.h>
#include <coreinit/mutex.h>
#include <coreinit/thread.h>
#include <sndcore2/core.h>
#include <sndcore2/voice.h>
#include <sndcore2/drcvs.h>
#define AUDIO_BUFFERS 2 // Double-buffer
#define MIN(a, b) ((a) < (b) ? (a) : (b))
#define MAX(a, b) ((a) > (b) ? (a) : (b))
#define CLAMP(x, y, z) MIN(MAX((x), (y)), (z))
static void (*parent_callback)(long *stream, size_t frames_total);
static OSMutex sound_list_mutex;
static OSMutex organya_mutex;
static AXVoice *voices[2];
static short *stream_buffers[2];
static long *stream_buffer_long;
static size_t buffer_length;
static void FrameCallback(void)
{
// We use a double-buffer: while the Wii U is busy playing one half of the buffer, we update the other.
// The buffer is 10ms long in total, and this function runs every 3ms.
// Just assume both voices are in-sync, and only check the first one
AXVoiceOffsets offsets;
AXGetVoiceOffsets(voices[0], &offsets);
unsigned int current_buffer = offsets.currentOffset / buffer_length;
static unsigned int last_buffer = 1;
if (current_buffer != last_buffer)
{
// Clear the mixer buffer
memset(stream_buffer_long, 0, buffer_length * sizeof(long) * 2);
// Fill mixer buffer
parent_callback(stream_buffer_long, buffer_length);
// Deinterlate samples, convert them to S16, and write them to the double-buffers
short *left_output_buffer = &stream_buffers[0][buffer_length * last_buffer];
short *right_output_buffer = &stream_buffers[1][buffer_length * last_buffer];
long *mixer_buffer_pointer = stream_buffer_long;
short *left_output_buffer_pointer = left_output_buffer;
short *right_output_buffer_pointer = right_output_buffer;
for (unsigned int i = 0; i < buffer_length; ++i)
{
const long left_sample = *mixer_buffer_pointer++;
const long right_sample = *mixer_buffer_pointer++;
// Clamp samples to sane limits, convert to S16, and store in double-buffers
if (left_sample > 0x7FFF)
*left_output_buffer_pointer++ = 0x7FFF;
else if (left_sample < -0x7FFF)
*left_output_buffer_pointer++ = -0x7FFF;
else
*left_output_buffer_pointer++ = (short)left_sample;
if (right_sample > 0x7FFF)
*right_output_buffer_pointer++ = 0x7FFF;
else if (right_sample < -0x7FFF)
*right_output_buffer_pointer++ = -0x7FFF;
else
*right_output_buffer_pointer++ = (short)right_sample;
}
// Make sure the sound hardware can see our data
DCStoreRange(left_output_buffer, buffer_length * sizeof(short));
DCStoreRange(right_output_buffer, buffer_length * sizeof(short));
last_buffer = current_buffer;
}
}
unsigned long SoftwareMixerBackend_Init(void (*callback)(long *stream, size_t frames_total))
{
if (!AXIsInit())
{
AXInitParams initparams = {
.renderer = AX_INIT_RENDERER_48KHZ,
.pipeline = AX_INIT_PIPELINE_SINGLE,
};
AXInitWithParams(&initparams);
}
OSInitMutex(&sound_list_mutex);
OSInitMutex(&organya_mutex);
unsigned long output_frequency = AXGetInputSamplesPerSec();
buffer_length = output_frequency / 100; // 10ms buffer
// Create and initialise two 'voices': each one will stream its own
// audio - one for the left speaker, and one for the right.
// The software-mixer outputs interlaced samples into a buffer of `long`s,
// so create a buffer for it here.
stream_buffer_long = (long*)malloc(buffer_length * sizeof(long) * 2); // `* 2` because it's an interlaced stereo buffer
if (stream_buffer_long != NULL)
{
stream_buffers[0] = (short*)malloc(buffer_length * sizeof(short) * AUDIO_BUFFERS);
if (stream_buffers[0] != NULL)
{
stream_buffers[1] = (short*)malloc(buffer_length * sizeof(short) * AUDIO_BUFFERS);
if (stream_buffers[1] != NULL)
{
voices[0] = AXAcquireVoice(31, NULL, NULL);
if (voices[0] != NULL)
{
voices[1] = AXAcquireVoice(31, NULL, NULL);
if (voices[1] != NULL)
{
for (unsigned int i = 0; i < 2; ++i)
{
AXVoiceBegin(voices[i]);
AXSetVoiceType(voices[i], 0);
AXVoiceVeData vol = {.volume = 0x8000};
AXSetVoiceVe(voices[i], &vol);
AXVoiceDeviceMixData mix_data[6];
memset(mix_data, 0, sizeof(mix_data));
mix_data[i].bus[0].volume = 0x8000; // Voice 1 goes on the left speaker - voice 2 goes on the right speaker
AXSetVoiceDeviceMix(voices[i], AX_DEVICE_TYPE_DRC, 0, mix_data);
AXSetVoiceDeviceMix(voices[i], AX_DEVICE_TYPE_TV, 0, mix_data);
AXSetVoiceSrcRatio(voices[i], 1.0f); // We use the native sample rate
AXSetVoiceSrcType(voices[i], AX_VOICE_SRC_TYPE_NONE);
AXVoiceOffsets offs = {
.dataType = AX_VOICE_FORMAT_LPCM16,
.loopingEnabled = AX_VOICE_LOOP_ENABLED,
.loopOffset = 0,
.endOffset = (buffer_length * AUDIO_BUFFERS) - 1, // -1 or else you'll get popping!
.currentOffset = 0,
.data = stream_buffers[i]
};
AXSetVoiceOffsets(voices[i], &offs);
AXVoiceEnd(voices[i]);
}
parent_callback = callback;
// Register the frame callback.
// Apparently, this fires every 3ms - we will use
// it to update the stream buffers when needed.
AXRegisterAppFrameCallback(FrameCallback);
return output_frequency;
}
AXFreeVoice(voices[0]);
}
free(stream_buffers[1]);
}
free(stream_buffers[0]);
}
free(stream_buffer_long);
}
AXQuit();
return 0;
}
void SoftwareMixerBackend_Deinit(void)
{
AXRegisterAppFrameCallback(NULL);
AXFreeVoice(voices[1]);
AXFreeVoice(voices[0]);
free(stream_buffers[1]);
free(stream_buffers[0]);
free(stream_buffer_long);
AXQuit();
}
bool SoftwareMixerBackend_Start(void)
{
AXSetVoiceState(voices[0], AX_VOICE_STATE_PLAYING);
AXSetVoiceState(voices[1], AX_VOICE_STATE_PLAYING);
return true;
}
void SoftwareMixerBackend_LockMixerMutex(void)
{
OSLockMutex(&sound_list_mutex);
}
void SoftwareMixerBackend_UnlockMixerMutex(void)
{
OSUnlockMutex(&sound_list_mutex);
}
void SoftwareMixerBackend_LockOrganyaMutex(void)
{
OSLockMutex(&organya_mutex);
}
void SoftwareMixerBackend_UnlockOrganyaMutex(void)
{
OSUnlockMutex(&organya_mutex);
}