mirror of
https://github.com/DrBeef/QuakeQuest.git
synced 2024-12-01 00:22:14 +00:00
605d1edb6e
- Audio lag is now fixed - using the OpenSLES library (courtesy of Emile Belanger's Dark Places build for Android) - 120hz mode can be used on the Quest 2 by adding the following to the command line: ``` -r 120 ```
399 lines
12 KiB
C
399 lines
12 KiB
C
/*
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===========================================================================
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Copyright (C) 1999-2005 Id Software, Inc.
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This file is part of Quake III Arena source code.
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Quake III Arena source code is free software; you can redistribute it
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and/or modify it under the terms of the GNU General Public License as
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published by the Free Software Foundation; either version 2 of the License,
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or (at your option) any later version.
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Quake III Arena source code is distributed in the hope that it will be
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useful, but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with Foobar; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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===========================================================================
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*/
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#include "quakedef.h"
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#include <limits.h>
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#include <pthread.h>
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#include <CoreAudio/AudioHardware.h>
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#include "snd_main.h"
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#define CHUNK_SIZE 1024
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static unsigned int submissionChunk = 0; // in sample frames
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static unsigned int coreaudiotime = 0; // based on the number of chunks submitted so far
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static qboolean s_isRunning = false;
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static pthread_mutex_t coreaudio_mutex;
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static AudioDeviceID outputDeviceID = kAudioDeviceUnknown;
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static short *mixbuffer = NULL;
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/*
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====================
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audioDeviceIOProc
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====================
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*/
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static OSStatus audioDeviceIOProc(AudioDeviceID inDevice,
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const AudioTimeStamp *inNow,
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const AudioBufferList *inInputData,
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const AudioTimeStamp *inInputTime,
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AudioBufferList *outOutputData,
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const AudioTimeStamp *inOutputTime,
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void *inClientData)
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{
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float *outBuffer;
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unsigned int frameCount, factor, sampleIndex;
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float scale = 1.0f / SHRT_MAX;
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outBuffer = (float*)outOutputData->mBuffers[0].mData;
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factor = snd_renderbuffer->format.channels * snd_renderbuffer->format.width;
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frameCount = 0;
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if (snd_blocked)
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scale = 0;
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// Lock the snd_renderbuffer
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if (SndSys_LockRenderBuffer())
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{
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unsigned int maxFrames, sampleCount;
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unsigned int startOffset, endOffset;
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const short *samples;
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if (snd_usethreadedmixing)
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{
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S_MixToBuffer(mixbuffer, submissionChunk);
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sampleCount = submissionChunk * snd_renderbuffer->format.channels;
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for (sampleIndex = 0; sampleIndex < sampleCount; sampleIndex++)
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outBuffer[sampleIndex] = mixbuffer[sampleIndex] * scale;
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// unlock the mutex now
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SndSys_UnlockRenderBuffer();
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return 0;
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}
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// Transfert up to a chunk of sample frames from snd_renderbuffer to outBuffer
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maxFrames = snd_renderbuffer->endframe - snd_renderbuffer->startframe;
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if (maxFrames >= submissionChunk)
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frameCount = submissionChunk;
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else
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frameCount = maxFrames;
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// Convert the samples from shorts to floats. Scale the floats to be [-1..1].
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startOffset = snd_renderbuffer->startframe % snd_renderbuffer->maxframes;
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endOffset = (snd_renderbuffer->startframe + frameCount) % snd_renderbuffer->maxframes;
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if (startOffset > endOffset) // if the buffer wraps
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{
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sampleCount = (snd_renderbuffer->maxframes - startOffset) * snd_renderbuffer->format.channels;
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samples = (const short*)(&snd_renderbuffer->ring[startOffset * factor]);
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for (sampleIndex = 0; sampleIndex < sampleCount; sampleIndex++)
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outBuffer[sampleIndex] = samples[sampleIndex] * scale;
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outBuffer = &outBuffer[sampleCount];
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sampleCount = frameCount * snd_renderbuffer->format.channels - sampleCount;
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samples = (const short*)(&snd_renderbuffer->ring[0]);
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for (sampleIndex = 0; sampleIndex < sampleCount; sampleIndex++)
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outBuffer[sampleIndex] = samples[sampleIndex] * scale;
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}
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else
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{
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sampleCount = frameCount * snd_renderbuffer->format.channels;
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samples = (const short*)(&snd_renderbuffer->ring[startOffset * factor]);
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for (sampleIndex = 0; sampleIndex < sampleCount; sampleIndex++)
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outBuffer[sampleIndex] = samples[sampleIndex] * scale;
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}
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snd_renderbuffer->startframe += frameCount;
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// unlock the mutex now
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SndSys_UnlockRenderBuffer();
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}
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// If there was not enough samples, complete with silence samples
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if (frameCount < submissionChunk)
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{
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unsigned int missingFrames;
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missingFrames = submissionChunk - frameCount;
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if (developer_insane.integer && vid_activewindow)
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Con_DPrintf("audioDeviceIOProc: %u sample frames missing\n", missingFrames);
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memset(&outBuffer[frameCount * snd_renderbuffer->format.channels], 0, missingFrames * sizeof(outBuffer[0]));
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}
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coreaudiotime += submissionChunk;
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return 0;
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}
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/*
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====================
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SndSys_Init
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Create "snd_renderbuffer" with the proper sound format if the call is successful
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May return a suggested format if the requested format isn't available
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====================
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*/
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qboolean SndSys_Init (const snd_format_t* requested, snd_format_t* suggested)
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{
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OSStatus status;
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UInt32 propertySize, bufferByteCount;
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AudioStreamBasicDescription streamDesc;
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if (s_isRunning)
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return true;
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Con_Printf("Initializing CoreAudio...\n");
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snd_threaded = false;
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if(requested->width != 2)
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{
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// we can only do 16bit per sample for now
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if(suggested != NULL)
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{
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memcpy (suggested, requested, sizeof (*suggested));
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suggested->width = 2;
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}
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return false;
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}
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// Get the output device
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propertySize = sizeof(outputDeviceID);
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status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propertySize, &outputDeviceID);
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if (status)
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{
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Con_Printf("CoreAudio: AudioDeviceGetProperty() returned %d when getting kAudioHardwarePropertyDefaultOutputDevice\n", (int)status);
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return false;
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}
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if (outputDeviceID == kAudioDeviceUnknown)
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{
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Con_Printf("CoreAudio: outputDeviceID is kAudioDeviceUnknown\n");
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return false;
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}
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// Configure the output device
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propertySize = sizeof(bufferByteCount);
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bufferByteCount = CHUNK_SIZE * sizeof(float) * requested->channels;
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status = AudioDeviceSetProperty(outputDeviceID, NULL, 0, false, kAudioDevicePropertyBufferSize, propertySize, &bufferByteCount);
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if (status)
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{
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Con_Printf("CoreAudio: AudioDeviceSetProperty() returned %d when setting kAudioDevicePropertyBufferSize to %d\n", (int)status, CHUNK_SIZE);
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return false;
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}
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propertySize = sizeof(bufferByteCount);
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status = AudioDeviceGetProperty(outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &bufferByteCount);
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if (status)
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{
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Con_Printf("CoreAudio: AudioDeviceGetProperty() returned %d when setting kAudioDevicePropertyBufferSize\n", (int)status);
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return false;
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}
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submissionChunk = bufferByteCount / sizeof(float);
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if (submissionChunk % requested->channels != 0)
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{
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Con_Print("CoreAudio: chunk size is NOT a multiple of the number of channels\n");
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return false;
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}
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submissionChunk /= requested->channels;
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Con_Printf(" Chunk size = %d sample frames\n", submissionChunk);
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// Print out the device status
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propertySize = sizeof(streamDesc);
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status = AudioDeviceGetProperty(outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &streamDesc);
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if (status)
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{
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Con_Printf("CoreAudio: AudioDeviceGetProperty() returned %d when getting kAudioDevicePropertyStreamFormat\n", (int)status);
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return false;
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}
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Con_Print (" Hardware format:\n");
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Con_Printf(" %5d mSampleRate\n", (unsigned int)streamDesc.mSampleRate);
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Con_Printf(" %c%c%c%c mFormatID\n",
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(char)(streamDesc.mFormatID >> 24),
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(char)(streamDesc.mFormatID >> 16),
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(char)(streamDesc.mFormatID >> 8),
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(char)(streamDesc.mFormatID >> 0));
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Con_Printf(" %5u mBytesPerPacket\n", (unsigned int)streamDesc.mBytesPerPacket);
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Con_Printf(" %5u mFramesPerPacket\n", (unsigned int)streamDesc.mFramesPerPacket);
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Con_Printf(" %5u mBytesPerFrame\n", (unsigned int)streamDesc.mBytesPerFrame);
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Con_Printf(" %5u mChannelsPerFrame\n", (unsigned int)streamDesc.mChannelsPerFrame);
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Con_Printf(" %5u mBitsPerChannel\n", (unsigned int)streamDesc.mBitsPerChannel);
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// Suggest proper settings if they differ
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if (requested->channels != streamDesc.mChannelsPerFrame || requested->speed != streamDesc.mSampleRate)
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{
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if (suggested != NULL)
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{
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memcpy (suggested, requested, sizeof (*suggested));
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suggested->channels = streamDesc.mChannelsPerFrame;
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suggested->speed = streamDesc.mSampleRate;
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}
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return false;
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}
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if(streamDesc.mFormatID == kAudioFormatLinearPCM)
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{
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// Add the callback function
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status = AudioDeviceAddIOProc(outputDeviceID, audioDeviceIOProc, NULL);
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if (!status)
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{
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// We haven't sent any sample frames yet
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coreaudiotime = 0;
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if (pthread_mutex_init(&coreaudio_mutex, NULL) == 0)
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{
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if ((snd_renderbuffer = Snd_CreateRingBuffer(requested, 0, NULL)))
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{
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if ((mixbuffer = Mem_Alloc(snd_mempool, CHUNK_SIZE * sizeof(*mixbuffer) * requested->channels)))
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{
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// Start sound running
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status = AudioDeviceStart(outputDeviceID, audioDeviceIOProc);
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if (!status)
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{
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s_isRunning = true;
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snd_threaded = true;
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Con_Print(" Initialization successful\n");
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return true;
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}
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else
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Con_Printf("CoreAudio: AudioDeviceStart() returned %d\n", (int)status);
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Mem_Free(mixbuffer);
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mixbuffer = NULL;
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}
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else
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Con_Print("CoreAudio: can't allocate memory for mixbuffer\n");
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Mem_Free(snd_renderbuffer->ring);
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Mem_Free(snd_renderbuffer);
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snd_renderbuffer = NULL;
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}
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else
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Con_Print("CoreAudio: can't allocate memory for ringbuffer\n");
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pthread_mutex_destroy(&coreaudio_mutex);
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}
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else
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Con_Print("CoreAudio: can't create pthread mutex\n");
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AudioDeviceRemoveIOProc(outputDeviceID, audioDeviceIOProc);
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}
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else
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Con_Printf("CoreAudio: AudioDeviceAddIOProc() returned %d\n", (int)status);
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}
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else
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Con_Print("CoreAudio: Default audio device doesn't support linear PCM!\n");
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return false;
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}
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/*
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====================
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SndSys_Shutdown
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Stop the sound card, delete "snd_renderbuffer" and free its other resources
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====================
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*/
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void SndSys_Shutdown(void)
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{
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OSStatus status;
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if (!s_isRunning)
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return;
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status = AudioDeviceStop(outputDeviceID, audioDeviceIOProc);
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if (status)
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{
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Con_Printf("AudioDeviceStop: returned %d\n", (int)status);
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return;
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}
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s_isRunning = false;
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pthread_mutex_destroy(&coreaudio_mutex);
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status = AudioDeviceRemoveIOProc(outputDeviceID, audioDeviceIOProc);
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if (status)
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{
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Con_Printf("AudioDeviceRemoveIOProc: returned %d\n", (int)status);
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return;
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}
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if (snd_renderbuffer != NULL)
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{
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Mem_Free(snd_renderbuffer->ring);
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Mem_Free(snd_renderbuffer);
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snd_renderbuffer = NULL;
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}
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if (mixbuffer != NULL)
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Mem_Free(mixbuffer);
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mixbuffer = NULL;
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}
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/*
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====================
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SndSys_Submit
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Submit the contents of "snd_renderbuffer" to the sound card
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====================
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*/
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void SndSys_Submit (void)
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{
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// Nothing to do here (this sound module is callback-based)
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}
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/*
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====================
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SndSys_GetSoundTime
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Returns the number of sample frames consumed since the sound started
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====================
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*/
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unsigned int SndSys_GetSoundTime (void)
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{
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return coreaudiotime;
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}
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/*
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====================
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SndSys_LockRenderBuffer
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Get the exclusive lock on "snd_renderbuffer"
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====================
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*/
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qboolean SndSys_LockRenderBuffer (void)
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{
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return (pthread_mutex_lock(&coreaudio_mutex) == 0);
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}
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/*
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====================
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SndSys_UnlockRenderBuffer
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Release the exclusive lock on "snd_renderbuffer"
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====================
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*/
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void SndSys_UnlockRenderBuffer (void)
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{
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pthread_mutex_unlock(&coreaudio_mutex);
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}
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/*
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====================
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SndSys_SendKeyEvents
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Send keyboard events originating from the sound system (e.g. MIDI)
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====================
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*/
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void SndSys_SendKeyEvents(void)
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{
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// not supported
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}
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