mirror of
https://github.com/moonlight-stream/moonlight-ios.git
synced 2026-06-16 21:50:57 +00:00
Switch to SDL for audio output
This commit is contained in:
Cameron Gutman
+64
-170
@@ -9,10 +9,11 @@
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#import "Connection.h"
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#import "Utils.h"
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#import <AudioUnit/AudioUnit.h>
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#import <AVFoundation/AVFoundation.h>
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#import <VideoToolbox/VideoToolbox.h>
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#define SDL_MAIN_HANDLED
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#import <SDL.h>
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#include "Limelight.h"
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#include "opus_multistream.h"
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@@ -37,24 +38,11 @@ static video_stats_t currentVideoStats;
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static video_stats_t lastVideoStats;
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static NSLock* videoStatsLock;
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#define OUTPUT_BUS 0
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static SDL_AudioDeviceID audioDevice;
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static OPUS_MULTISTREAM_CONFIGURATION audioConfig;
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static void* audioBuffer;
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static int audioFrameSize;
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// My iPod touch 5th Generation seems to really require 80 ms
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// of buffering to deliver glitch-free playback :(
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// FIXME: Maybe we can use a smaller buffer on more modern iOS versions?
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#define CIRCULAR_BUFFER_DURATION 80
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static int audioBufferEntries;
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static int audioBufferWriteIndex;
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static int audioBufferReadIndex;
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static int audioBufferStride;
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static int audioSamplesPerFrame;
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static short* audioCircularBuffer;
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#define AUDIO_QUEUE_BUFFERS 3
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static AudioQueueRef audioQueue;
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static AudioQueueBufferRef audioBuffers[AUDIO_QUEUE_BUFFERS];
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static VideoDecoderRenderer* renderer;
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int DrDecoderSetup(int videoFormat, int width, int height, int redrawRate, void* context, int drFlags)
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@@ -169,113 +157,49 @@ int DrSubmitDecodeUnit(PDECODE_UNIT decodeUnit)
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pts:decodeUnit->presentationTimeMs];
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}
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int ArInit(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION originalOpusConfig, void* context, int flags)
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int ArInit(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig, void* context, int flags)
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{
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int err;
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AudioChannelLayout channelLayout = {};
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OPUS_MULTISTREAM_CONFIGURATION opusConfig = *originalOpusConfig;
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SDL_AudioSpec want, have;
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// Initialize the circular buffer
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audioBufferWriteIndex = audioBufferReadIndex = 0;
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audioSamplesPerFrame = opusConfig.samplesPerFrame;
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audioBufferStride = opusConfig.channelCount * opusConfig.samplesPerFrame;
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audioBufferEntries = CIRCULAR_BUFFER_DURATION / (opusConfig.samplesPerFrame / (opusConfig.sampleRate / 1000));
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audioCircularBuffer = malloc(audioBufferEntries * audioBufferStride * sizeof(short));
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if (audioCircularBuffer == NULL) {
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Log(LOG_E, @"Error allocating output queue\n");
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
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Log(LOG_E, @"Failed to initialize audio subsystem: %s\n", SDL_GetError());
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return -1;
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}
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SDL_zero(want);
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want.freq = opusConfig->sampleRate;
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want.format = AUDIO_S16;
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want.channels = opusConfig->channelCount;
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// This is supposed to be a power of 2, but our
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// frames contain a non-power of 2 number of samples,
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// so the slop would require buffering another full frame.
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// Specifying non-Po2 seems to work for our supported platforms.
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want.samples = opusConfig->samplesPerFrame;
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audioDevice = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
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if (audioDevice == 0) {
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Log(LOG_E, @"Failed to open audio device: %s\n", SDL_GetError());
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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return -1;
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}
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switch (opusConfig.channelCount) {
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case 2:
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channelLayout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
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break;
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case 4:
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channelLayout.mChannelLayoutTag = kAudioChannelLayoutTag_Quadraphonic;
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break;
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case 6:
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channelLayout.mChannelLayoutTag = kAudioChannelLayoutTag_AudioUnit_5_1;
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break;
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case 8:
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channelLayout.mChannelLayoutTag = kAudioChannelLayoutTag_AudioUnit_7_1;
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// Swap SL/SR and RL/RR to match the selected channel layout
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opusConfig.mapping[4] = originalOpusConfig->mapping[6];
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opusConfig.mapping[5] = originalOpusConfig->mapping[7];
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opusConfig.mapping[6] = originalOpusConfig->mapping[4];
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opusConfig.mapping[7] = originalOpusConfig->mapping[5];
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break;
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default:
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// Unsupported channel layout
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Log(LOG_E, @"Unsupported channel layout: %d\n", opusConfig.channelCount);
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abort();
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}
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audioConfig = *opusConfig;
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audioFrameSize = opusConfig->samplesPerFrame * sizeof(short) * opusConfig->channelCount;
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audioBuffer = SDL_malloc(audioFrameSize);
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opusDecoder = opus_multistream_decoder_create(opusConfig.sampleRate,
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opusConfig.channelCount,
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opusConfig.streams,
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opusConfig.coupledStreams,
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opusConfig.mapping,
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opusDecoder = opus_multistream_decoder_create(opusConfig->sampleRate,
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opusConfig->channelCount,
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opusConfig->streams,
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opusConfig->coupledStreams,
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opusConfig->mapping,
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&err);
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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:opusConfig.sampleRate error:&audioSessionError];
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[audioSession setCategory:AVAudioSessionCategoryPlayback
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withOptions:AVAudioSessionCategoryOptionMixWithOthers
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error:&audioSessionError];
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[audioSession setPreferredIOBufferDuration:(opusConfig.samplesPerFrame / (opusConfig.sampleRate / 1000)) / 1000.0
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error:&audioSessionError];
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[audioSession setActive: YES error: &audioSessionError];
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// FIXME: Calling this breaks surround audio for some reason
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//[audioSession setPreferredOutputNumberOfChannels:opusConfig->channelCount error:&audioSessionError];
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// Start playback
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SDL_PauseAudioDevice(audioDevice, 0);
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OSStatus status;
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AudioStreamBasicDescription audioFormat = {0};
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audioFormat.mSampleRate = opusConfig.sampleRate;
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audioFormat.mBitsPerChannel = 16;
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audioFormat.mFormatID = kAudioFormatLinearPCM;
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audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
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audioFormat.mChannelsPerFrame = opusConfig.channelCount;
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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 = AudioQueueNewOutput(&audioFormat, FillOutputBuffer, nil, nil, nil, 0, &audioQueue);
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if (status != noErr) {
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Log(LOG_E, @"Error allocating output queue: %d\n", status);
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return status;
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}
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// We need to specify a channel layout for surround sound configurations
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status = AudioQueueSetProperty(audioQueue, kAudioQueueProperty_ChannelLayout, &channelLayout, sizeof(channelLayout));
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if (status != noErr) {
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Log(LOG_E, @"Error configuring surround channel layout: %d\n", status);
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return status;
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}
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for (int i = 0; i < AUDIO_QUEUE_BUFFERS; i++) {
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status = AudioQueueAllocateBuffer(audioQueue, audioFormat.mBytesPerFrame * opusConfig.samplesPerFrame, &audioBuffers[i]);
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if (status != noErr) {
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Log(LOG_E, @"Error allocating output buffer: %d\n", status);
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return status;
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}
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FillOutputBuffer(nil, audioQueue, audioBuffers[i]);
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}
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status = AudioQueueStart(audioQueue, nil);
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if (status != noErr) {
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Log(LOG_E, @"Error starting queue: %d\n", status);
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return status;
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}
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return status;
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return 0;
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}
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void ArCleanup(void)
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@@ -285,43 +209,43 @@ void ArCleanup(void)
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opusDecoder = NULL;
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}
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// Stop before disposing to avoid massive delay inside
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// AudioQueueDispose() (iOS bug?)
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AudioQueueStop(audioQueue, true);
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// Also frees buffers
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AudioQueueDispose(audioQueue, true);
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// Must be freed after the queue is stopped
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if (audioCircularBuffer != NULL) {
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free(audioCircularBuffer);
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audioCircularBuffer = NULL;
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if (audioDevice != 0) {
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SDL_CloseAudioDevice(audioDevice);
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audioDevice = 0;
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}
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// Audio session is now inactive
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[[AVAudioSession sharedInstance] setActive: NO error: nil];
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if (audioBuffer != NULL) {
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SDL_free(audioBuffer);
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audioBuffer = NULL;
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}
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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}
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void ArDecodeAndPlaySample(char* sampleData, int sampleLength)
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{
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int decodeLen;
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// Check if there is space for this sample in the buffer. Again, this can race
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// but in the worst case, we'll not see the sample callback having consumed a sample.
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if (((audioBufferWriteIndex + 1) % audioBufferEntries) == audioBufferReadIndex) {
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// Don't queue if there's already more than 30 ms of audio data waiting
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// in Moonlight's audio queue.
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if (LiGetPendingAudioDuration() > 30) {
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return;
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}
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decodeLen = opus_multistream_decode(opusDecoder, (unsigned char *)sampleData, sampleLength,
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(short*)&audioCircularBuffer[audioBufferWriteIndex * audioBufferStride], audioSamplesPerFrame, 0);
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(short*)audioBuffer, audioConfig.samplesPerFrame, 0);
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if (decodeLen > 0) {
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// Use a full memory barrier to ensure the circular buffer is written before incrementing the index
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__sync_synchronize();
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// Provide backpressure on the queue to ensure too many frames don't build up
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// in SDL's audio queue.
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while (SDL_GetQueuedAudioSize(audioDevice) / audioFrameSize > 10) {
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SDL_Delay(1);
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}
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// This can race with the reader in the sample callback, however this is a benign
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// race since we'll either read the original value of s_WriteIndex (which is safe,
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// we just won't consider this sample) or the new value of s_WriteIndex
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audioBufferWriteIndex = (audioBufferWriteIndex + 1) % audioBufferEntries;
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if (SDL_QueueAudio(audioDevice,
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audioBuffer,
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sizeof(short) * decodeLen * audioConfig.channelCount) < 0) {
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Log(LOG_E, @"Failed to queue audio sample: %s\n", SDL_GetError());
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}
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}
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}
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@@ -508,8 +432,7 @@ void ClConnectionStatusUpdate(int status)
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_arCallbacks.init = ArInit;
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_arCallbacks.cleanup = ArCleanup;
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_arCallbacks.decodeAndPlaySample = ArDecodeAndPlaySample;
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_arCallbacks.capabilities = CAPABILITY_DIRECT_SUBMIT |
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CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
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_arCallbacks.capabilities = CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION;
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LiInitializeConnectionCallbacks(&_clCallbacks);
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_clCallbacks.stageStarting = ClStageStarting;
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@@ -524,35 +447,6 @@ void ClConnectionStatusUpdate(int status)
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return self;
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}
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static void FillOutputBuffer(void *aqData,
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AudioQueueRef inAQ,
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AudioQueueBufferRef inBuffer) {
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inBuffer->mAudioDataByteSize = audioBufferStride * sizeof(short);
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assert(inBuffer->mAudioDataByteSize == inBuffer->mAudioDataBytesCapacity);
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// If the indexes aren't equal, we have a sample
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if (audioBufferWriteIndex != audioBufferReadIndex) {
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// Copy data to the audio buffer
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memcpy(inBuffer->mAudioData,
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&audioCircularBuffer[audioBufferReadIndex * audioBufferStride],
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inBuffer->mAudioDataByteSize);
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// Use a full memory barrier to ensure the circular buffer is read before incrementing the index
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__sync_synchronize();
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// This can race with the reader in the AudDecDecodeAndPlaySample function. This is
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// not a problem because at worst, it just won't see that we've consumed this sample yet.
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audioBufferReadIndex = (audioBufferReadIndex + 1) % audioBufferEntries;
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}
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else {
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// No data, so play silence
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memset(inBuffer->mAudioData, 0, inBuffer->mAudioDataByteSize);
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}
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AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
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}
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-(void) main
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{
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[initLock lock];
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