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https://github.com/moonlight-stream/moonlight-ios.git
synced 2025-07-22 12:13:35 +00:00
Implement working audio support
This commit is contained in:
Cameron Gutman
parent
93063f5606
commit
70f3a91dfb
@ -28,10 +28,17 @@ static OpusDecoder *opusDecoder;
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#define PCM_BUFFER_SIZE 1024
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#define OUTPUT_BUS 0
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static short* decodedPcmBuffer;
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static int filledPcmBuffer;
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struct AUDIO_BUFFER_QUEUE_ENTRY {
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struct AUDIO_BUFFER_QUEUE_ENTRY *next;
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int length;
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int offset;
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char data[0];
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};
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static short decodedPcmBuffer[512];
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static NSLock *audioLock;
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static struct AUDIO_BUFFER_QUEUE_ENTRY *audioBufferQueue;
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static AudioComponentInstance audioUnit;
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static bool started = false;
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static VideoDecoderRenderer* renderer;
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void DrSetup(int width, int height, int fps, void* context, int drFlags)
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@ -81,18 +88,13 @@ void ArInit(void)
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opusDecoder = opus_decoder_create(48000, 2, &err);
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decodedPcmBuffer = malloc(PCM_BUFFER_SIZE);
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audioLock = [[NSLock alloc] init];
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}
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void ArRelease(void)
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{
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printf("Release audio\n");
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if (decodedPcmBuffer != NULL) {
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free(decodedPcmBuffer);
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decodedPcmBuffer = NULL;
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}
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if (opusDecoder != NULL) {
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opus_decoder_destroy(opusDecoder);
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opusDecoder = NULL;
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@ -102,6 +104,7 @@ void ArRelease(void)
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void ArStart(void)
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{
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printf("Start audio\n");
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AudioOutputUnitStart(audioUnit);
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}
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void ArStop(void)
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@ -111,17 +114,31 @@ void ArStop(void)
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void ArDecodeAndPlaySample(char* sampleData, int sampleLength)
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{
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if (!started) {
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AudioOutputUnitStart(audioUnit);
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started = true;
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}
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filledPcmBuffer = opus_decode(opusDecoder, (unsigned char*)sampleData, sampleLength, decodedPcmBuffer, PCM_BUFFER_SIZE / 2, 0);
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if (filledPcmBuffer > 0) {
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int decodedLength = opus_decode(opusDecoder, (unsigned char*)sampleData, sampleLength, decodedPcmBuffer, PCM_BUFFER_SIZE / 2, 0);
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if (decodedLength > 0) {
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// Return of opus_decode is samples per channel
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filledPcmBuffer *= 4;
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decodedLength *= 4;
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NSLog(@"pcmBuffer: %d", filledPcmBuffer);
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struct AUDIO_BUFFER_QUEUE_ENTRY *newEntry = malloc(sizeof(*newEntry) + decodedLength);
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if (newEntry != NULL) {
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newEntry->next = NULL;
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newEntry->length = decodedLength;
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newEntry->offset = 0;
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memcpy(newEntry->data, decodedPcmBuffer, decodedLength);
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[audioLock lock];
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if (audioBufferQueue == NULL) {
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audioBufferQueue = newEntry;
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}
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else {
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struct AUDIO_BUFFER_QUEUE_ENTRY *lastEntry = audioBufferQueue;
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while (lastEntry->next != NULL) {
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lastEntry = lastEntry->next;
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}
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lastEntry->next = newEntry;
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}
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[audioLock unlock];
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}
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}
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}
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@ -193,15 +210,15 @@ void ClDisplayTransientMessage(char* message)
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clCallbacks.displayMessage = ClDisplayMessage;
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clCallbacks.displayTransientMessage = ClDisplayTransientMessage;
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//////// Don't think any of this is used /////////
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// Configure the audio session for our app
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NSError *audioSessionError = nil;
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AVAudioSession* audioSession = [AVAudioSession sharedInstance];
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[audioSession setPreferredSampleRate:48000.0 error:&audioSessionError];
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[audioSession setPreferredSampleRate:48000.0 error:&audioSessionError];
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[audioSession setCategory: AVAudioSessionCategoryPlayAndRecord error: &audioSessionError];
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[audioSession setPreferredOutputNumberOfChannels:2 error:&audioSessionError];
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[audioSession setPreferredIOBufferDuration:0.005 error:&audioSessionError];
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[audioSession setActive: YES error: &audioSessionError];
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//////////////////////////////////////////////////
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OSStatus status;
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@ -217,27 +234,26 @@ void ClDisplayTransientMessage(char* message)
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if (status) {
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NSLog(@"Unable to instantiate new AudioComponent: %d", (int32_t)status);
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}
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AudioStreamBasicDescription audioFormat = {0};
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audioFormat.mSampleRate = 48000;
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audioFormat.mBitsPerChannel = 16;
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audioFormat.mFormatID = kAudioFormatLinearPCM;
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audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger;
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audioFormat.mFramesPerPacket = 1;
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audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
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audioFormat.mChannelsPerFrame = 2;
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audioFormat.mBytesPerFrame = 960;
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audioFormat.mBytesPerPacket = 960;
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audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame * (audioFormat.mBitsPerChannel / 8);
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audioFormat.mBytesPerPacket = audioFormat.mBytesPerFrame;
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audioFormat.mFramesPerPacket = audioFormat.mBytesPerPacket / audioFormat.mBytesPerFrame;
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audioFormat.mReserved = 0;
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status = AudioUnitSetProperty(audioUnit,
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kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Output,
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kAudioUnitScope_Input,
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OUTPUT_BUS,
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&audioFormat,
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sizeof(audioFormat));
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if (status) {
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NSLog(@"Unable to set audio unit to output: %d", (int32_t)status);
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NSLog(@"Unable to set audio unit to input: %d", (int32_t)status);
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}
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AURenderCallbackStruct callbackStruct = {0};
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@ -246,7 +262,7 @@ void ClDisplayTransientMessage(char* message)
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status = AudioUnitSetProperty(audioUnit,
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kAudioUnitProperty_SetRenderCallback,
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kAudioUnitScope_Global,
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kAudioUnitScope_Input,
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OUTPUT_BUS,
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&callbackStruct,
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sizeof(callbackStruct));
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@ -272,15 +288,60 @@ static OSStatus playbackCallback(void *inRefCon,
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// Fill them up as much as you can. Remember to set the size value in each buffer to match how
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// much data is in the buffer.
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NSLog(@"Playback callback");
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for (int i = 0; i < ioData->mNumberBuffers; i++) {
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ioData->mBuffers[i].mNumberChannels = 2;
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int min = MIN(ioData->mBuffers[i].mDataByteSize, filledPcmBuffer);
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NSLog(@"Min: %d", min);
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memcpy(ioData->mBuffers[i].mData, decodedPcmBuffer, min);
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ioData->mBuffers[i].mDataByteSize = min;
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filledPcmBuffer -= min;
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if (ioData->mBuffers[i].mDataByteSize != 0) {
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int thisBufferOffset = 0;
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FillBufferAgain:
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// Make sure there's data to write
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if (ioData->mBuffers[i].mDataByteSize - thisBufferOffset == 0) {
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continue;
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}
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// Wait for a buffer to be available
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// FIXME: This needs optimization to avoid busy waiting for buffers
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struct AUDIO_BUFFER_QUEUE_ENTRY *audioEntry = NULL;
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while (audioEntry == NULL)
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{
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[audioLock lock];
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if (audioBufferQueue != NULL) {
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// Dequeue this entry temporarily
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audioEntry = audioBufferQueue;
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audioBufferQueue = audioBufferQueue->next;
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}
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[audioLock unlock];
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}
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// Figure out how much data we can write
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int min = MIN(ioData->mBuffers[i].mDataByteSize - thisBufferOffset, audioEntry->length);
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// Copy data to the audio buffer
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memcpy(&ioData->mBuffers[i].mData[thisBufferOffset], &audioEntry->data[audioEntry->offset], min);
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thisBufferOffset += min;
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if (min < audioEntry->length) {
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// This entry still has unused data
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audioEntry->length -= min;
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audioEntry->offset += min;
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// Requeue the entry
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[audioLock lock];
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audioEntry->next = audioBufferQueue;
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audioBufferQueue = audioEntry;
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[audioLock unlock];
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}
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else {
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// This entry is fully depleted so free it
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free(audioEntry);
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// Try to grab another sample to fill this buffer with
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goto FillBufferAgain;
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}
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ioData->mBuffers[i].mDataByteSize = thisBufferOffset;
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}
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}
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return noErr;
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