mirror of
https://github.com/RetroDECK/Supermodel.git
synced 2024-11-26 23:55:40 +00:00
294 lines
8.2 KiB
C++
294 lines
8.2 KiB
C++
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#include "Supermodel.h"
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#ifdef SUPERMODEL_OSX
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#include <SDL/SDL.h>
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#include <SDL/SDL_audio.h>
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#else
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#include <SDL.h>
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#include <SDL_audio.h>
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#endif
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#include <math.h>
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// Model3 audio output is 44.1KHz 2-channel sound and frame rate is 60fps
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#define SAMPLE_RATE 44100
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#define NUM_CHANNELS 2
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#define SUPERMODEL_FPS 60
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#define BYTES_PER_SAMPLE (NUM_CHANNELS * sizeof(INT16))
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#define SAMPLES_PER_FRAME (SAMPLE_RATE / SUPERMODEL_FPS)
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#define BYTES_PER_FRAME (SAMPLES_PER_FRAME * BYTES_PER_SAMPLE)
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#define MAX_LATENCY 100
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static bool enabled = true; // True if sound output is enabled
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static unsigned latency = 20; // Audio latency to use (ie size of audio buffer) as percentage of max buffer size
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static unsigned playSamples = 512; // Size (in samples) of callback play buffer
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static UINT32 audioBufferSize = 0; // Size (in bytes) of audio buffer
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static INT8 *audioBuffer = NULL; // Audio buffer
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static UINT32 writePos = 0; // Current position at which writing into buffer
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static UINT32 playPos = 0; // Current position at which playing data in buffer via callback
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static bool writeWrapped = false; // True if write position has wrapped around at end of buffer but play position has not done so yet
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static unsigned underRuns = 0; // Number of buffer under-runs that have occured
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static unsigned overRuns = 0; // Number of buffer over-runs that have occured
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static void PlayCallback(void *data, Uint8 *stream, int len)
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{
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//printf("PlayCallback(%d)\n", len);
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// Get current write position and adjust it if write has wrapped
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UINT32 adjWritePos = writePos;
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if (writeWrapped)
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adjWritePos += audioBufferSize;
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// Check if play position overlaps write position (ie buffer under-run)
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if (adjWritePos < playPos + len)
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{
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// If so, just copy silence to audio output stream and exit
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memset(stream, 0, len);
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//printf("Audio buffer under-run in PlayCallback\n");
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underRuns++;
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return;
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}
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INT8* src1;
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INT8* src2;
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UINT32 len1;
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UINT32 len2;
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// Check if play region extends past end of buffer
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if (playPos + len > audioBufferSize)
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{
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// If so, split play region into two
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src1 = audioBuffer + playPos;
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src2 = audioBuffer;
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len1 = audioBufferSize - playPos;
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len2 = len - len1;
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}
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else
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{
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// Otherwise, just copy whole region
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src1 = audioBuffer + playPos;
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src2 = 0;
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len1 = len;
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len2 = 0;
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}
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// Check if audio is enabled
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if (enabled)
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{
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// If so, copy play region into audio output stream and blank region out afterwards
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memcpy(stream, src1, len1);
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memset(src1, 0, len1);
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if (len2)
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{
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// If region was split into two, copy second half into audio output stream as well and blank region out afterwards
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memcpy(stream + len1, src2, len2);
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memset(src2, 0, len2);
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}
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}
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else
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// Otherwise, just copy silence to audio output stream
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memset(stream, 0, len);
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// Move play position forward for next time
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playPos += len;
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// Check if play position has moved past end of buffer
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if (playPos >= audioBufferSize)
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{
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// If so, wrap it around to beginning again and reset write wrap flag
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playPos -= audioBufferSize;
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writeWrapped = false;
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}
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}
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static void MixChannels(unsigned numSamples, INT16 *leftBuffer, INT16 *rightBuffer, void *dest)
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{
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INT16 *p = (INT16*)dest;
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for (unsigned i = 0; i < numSamples; i++)
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{
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#if (NUM_CHANNELS == 1)
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*p++ = leftBuffer[i] + rightBuffer[i];
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#else
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*p++ = rightBuffer[i];
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*p++ = leftBuffer[i];
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#endif
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}
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}
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BOOL OpenAudio()
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{
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// Initialize SDL audio sub-system
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0)
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{
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ErrorLog("Unable to initialize SDL audio sub-system: %s\n", SDL_GetError());
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return FAIL;
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}
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// Set up audio specification
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SDL_AudioSpec fmt;
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memset(&fmt, 0, sizeof(SDL_AudioSpec));
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fmt.freq = SAMPLE_RATE;
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fmt.channels = NUM_CHANNELS;
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fmt.format = AUDIO_S16SYS;
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fmt.samples = playSamples;
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fmt.callback = PlayCallback;
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// Try opening SDL audio output with that specification
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SDL_AudioSpec obtained;
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if (SDL_OpenAudio(&fmt, &obtained) < 0)
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{
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ErrorLog("Unable to open 44.1KHz 2-channel audio with SDL: %s\n", SDL_GetError());
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return FAIL;
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}
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// Check what buffer sample size was actually obtained, and use that
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playSamples = obtained.samples;
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// Create audio buffer
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audioBufferSize = SAMPLE_RATE * BYTES_PER_SAMPLE * latency / MAX_LATENCY;
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int roundBuffer = 2 * playSamples;
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audioBufferSize = max<int>(roundBuffer, (audioBufferSize / roundBuffer) * roundBuffer);
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audioBuffer = new INT8[audioBufferSize];
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memset(audioBuffer, 0, sizeof(INT8) * audioBufferSize);
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// Set initial play position to be beginning of buffer and initial write position to be half-way into buffer
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playPos = 0;
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writePos = (BYTES_PER_FRAME + audioBufferSize) / 2;
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writeWrapped = false;
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// Reset counters
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underRuns = 0;
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overRuns = 0;
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// Start audio playing
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SDL_PauseAudio(0);
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return OKAY;
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}
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void OutputAudio(unsigned numSamples, INT16 *leftBuffer, INT16 *rightBuffer)
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{
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//printf("OutputAudio(%u)\n", numSamples);
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// Number of samples should never be more than max number of samples per frame
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if (numSamples > SAMPLES_PER_FRAME)
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numSamples = SAMPLES_PER_FRAME;
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// Mix together left and right channels into single chunk of data
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INT16 mixBuffer[NUM_CHANNELS * SAMPLES_PER_FRAME];
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MixChannels(numSamples, leftBuffer, rightBuffer, mixBuffer);
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// Lock SDL audio callback so that it doesn't interfere with following code
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SDL_LockAudio();
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// Calculate number of bytes for current sound chunk
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UINT32 numBytes = numSamples * BYTES_PER_SAMPLE;
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// Get end of current play region (writing must occur past this point)
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UINT32 playEndPos = playPos + BYTES_PER_FRAME;
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// Undo any wrap-around of the write position that may have occured to create following ordering: playPos < playEndPos < writePos
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if (writePos < playEndPos && writeWrapped)
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writePos += audioBufferSize;
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// If play region has caught up with write position and now overlaps it (ie buffer under-run), then bump write position forward in chunks
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// until it is past end of play region
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while (writePos < playEndPos)
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{
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//printf("Audio buffer under-run in OutputAudio\n");
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underRuns++;
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writePos += numBytes;
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}
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// If write position has caught up with play region and now overlaps it (ie buffer over-run), then discard current chunk of data
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if (writePos + numBytes > playPos + audioBufferSize)
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{
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//printf("Audio buffer over-run in OutputAudio\n");
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overRuns++;
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goto Finish;
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}
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// Check if write position has moved past end of buffer
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if (writePos >= audioBufferSize)
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{
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// If so, wrap it around to beginning again and set write wrap flag
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writePos -= audioBufferSize;
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writeWrapped = true;
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}
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INT16 *src = mixBuffer;
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INT8 *dst1;
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INT8 *dst2;
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UINT32 len1;
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UINT32 len2;
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// Check if write region extends past end of buffer
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if (writePos + numBytes > audioBufferSize)
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{
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// If so, split write region into two
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dst1 = audioBuffer + writePos;
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dst2 = audioBuffer;
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len1 = audioBufferSize - writePos;
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len2 = numBytes - len1;
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}
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else
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{
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// Otherwise, just copy whole region
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dst1 = audioBuffer + writePos;
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dst2 = 0;
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len1 = numBytes;
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len2 = 0;
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}
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// Copy chunk to write position in buffer
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UINT32 bytesRemaining = numBytes;
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UINT32 bytesToCopy = (bytesRemaining > len1 ? len1 : bytesRemaining);
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memcpy(dst1, src, bytesToCopy);
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// Adjust for number of bytes copied
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bytesRemaining -= bytesToCopy;
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src = (INT16*)((UINT8*)src + bytesToCopy);
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if (bytesRemaining)
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{
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// If write region was split into two, copy second half of chunk into buffer as well
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bytesToCopy = (bytesRemaining > len2 ? len2 : bytesRemaining);
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memcpy(dst2, src, bytesToCopy);
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}
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// Move write position forward for next time
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writePos += numBytes;
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// Check if write position has moved past end of buffer
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if (writePos >= audioBufferSize)
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{
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// If so wrap it around to beginning again and set write wrap flag
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writePos -= audioBufferSize;
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writeWrapped = true;
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}
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Finish:
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// Unlock SDL audio callback
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SDL_UnlockAudio();
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}
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void CloseAudio()
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{
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// Close SDL audio output
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SDL_CloseAudio();
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// Delete audio buffer
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if (audioBuffer != NULL)
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{
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delete[] audioBuffer;
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audioBuffer = NULL;
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}
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}
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