mirror of
https://github.com/moonlight-stream/moonlight-ios.git
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Merge branch 'master' of github.com:limelight-stream/limelight-ios
This commit is contained in:
Cameron Gutman
417
Limelight/Stream/Connection.m
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417
Limelight/Stream/Connection.m
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//
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// Connection.m
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// Limelight-iOS
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//
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// Created by Diego Waxemberg on 1/19/14.
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// Copyright (c) 2014 Diego Waxemberg. All rights reserved.
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//
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#import "Connection.h"
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#import <AudioUnit/AudioUnit.h>
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#import <AVFoundation/AVFoundation.h>
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#include "Limelight.h"
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#include "opus.h"
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@implementation Connection {
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IP_ADDRESS _host;
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STREAM_CONFIGURATION _streamConfig;
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CONNECTION_LISTENER_CALLBACKS _clCallbacks;
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DECODER_RENDERER_CALLBACKS _drCallbacks;
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AUDIO_RENDERER_CALLBACKS _arCallbacks;
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}
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static OpusDecoder *opusDecoder;
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static id<ConTermCallback> _callback;
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#define PCM_BUFFER_SIZE 1024
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#define OUTPUT_BUS 0
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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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#define MAX_QUEUE_ENTRIES 10
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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 int audioBufferQueueLength;
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static AudioComponentInstance audioUnit;
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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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{
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}
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void DrSubmitDecodeUnit(PDECODE_UNIT decodeUnit)
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{
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unsigned char* data = (unsigned char*) malloc(decodeUnit->fullLength);
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if (data != NULL) {
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int offset = 0;
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PLENTRY entry = decodeUnit->bufferList;
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while (entry != NULL) {
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memcpy(&data[offset], entry->data, entry->length);
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offset += entry->length;
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entry = entry->next;
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}
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// This function will take our buffer
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[renderer submitDecodeBuffer:data length:decodeUnit->fullLength];
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}
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}
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void DrStart(void)
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{
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}
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void DrStop(void)
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{
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}
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void DrRelease(void)
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{
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}
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void ArInit(void)
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{
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int err;
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opusDecoder = opus_decoder_create(48000, 2, &err);
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audioLock = [[NSLock alloc] init];
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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 setCategory: AVAudioSessionCategoryPlayback 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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OSStatus status;
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AudioComponentDescription audioDesc;
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audioDesc.componentType = kAudioUnitType_Output;
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audioDesc.componentSubType = kAudioUnitSubType_RemoteIO;
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audioDesc.componentFlags = 0;
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audioDesc.componentFlagsMask = 0;
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audioDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
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status = AudioComponentInstanceNew(AudioComponentFindNext(NULL, &audioDesc), &audioUnit);
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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 | kLinearPCMFormatFlagIsPacked;
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audioFormat.mChannelsPerFrame = 2;
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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_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 input: %d", (int32_t)status);
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}
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AURenderCallbackStruct callbackStruct = {0};
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callbackStruct.inputProc = playbackCallback;
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callbackStruct.inputProcRefCon = NULL;
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status = AudioUnitSetProperty(audioUnit,
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kAudioUnitProperty_SetRenderCallback,
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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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if (status) {
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NSLog(@"Unable to set audio unit callback: %d", (int32_t)status);
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}
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status = AudioUnitInitialize(audioUnit);
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if (status) {
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NSLog(@"Unable to initialize audioUnit: %d", (int32_t)status);
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}
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}
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void ArRelease(void)
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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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}
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OSStatus status = AudioUnitUninitialize(audioUnit);
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if (status) {
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NSLog(@"Unable to uninitialize audioUnit: %d", (int32_t)status);
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}
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// Audio session is now inactive
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AVAudioSession* audioSession = [AVAudioSession sharedInstance];
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[audioSession setActive: YES error: nil];
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// This is safe because we're guaranteed that nobody
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// is touching this list now
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struct AUDIO_BUFFER_QUEUE_ENTRY *entry;
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while (audioBufferQueue != NULL) {
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entry = audioBufferQueue;
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audioBufferQueue = entry->next;
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audioBufferQueueLength--;
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free(entry);
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}
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}
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void ArStart(void)
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{
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OSStatus status = AudioOutputUnitStart(audioUnit);
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if (status) {
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NSLog(@"Unable to start audioUnit: %d", (int32_t)status);
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}
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}
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void ArStop(void)
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{
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OSStatus status = AudioOutputUnitStop(audioUnit);
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if (status) {
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NSLog(@"Unable to stop audioUnit: %d", (int32_t)status);
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}
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}
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void ArDecodeAndPlaySample(char* sampleData, int sampleLength)
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{
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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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decodedLength *= 4;
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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 (audioBufferQueueLength > MAX_QUEUE_ENTRIES) {
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NSLog(@"Audio player too slow. Dropping all decoded samples!");
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// Clear all values from the buffer queue
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struct AUDIO_BUFFER_QUEUE_ENTRY *entry;
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while (audioBufferQueue != NULL) {
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entry = audioBufferQueue;
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audioBufferQueue = entry->next;
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audioBufferQueueLength--;
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free(entry);
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}
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}
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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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audioBufferQueueLength++;
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[audioLock unlock];
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}
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}
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}
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void ClStageStarting(int stage)
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{
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}
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void ClStageComplete(int stage)
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{
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}
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void ClStageFailed(int stage, long errorCode)
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{
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NSLog(@"Stage %d failed: %ld", stage, errorCode);
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}
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void ClConnectionStarted(void)
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{
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NSLog(@"Connection started");
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}
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void ClConnectionTerminated(long errorCode)
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{
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NSLog(@"ConnectionTerminated: %ld", errorCode);
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[_callback connectionTerminated];
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}
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void ClDisplayMessage(char* message)
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{
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NSLog(@"DisplayMessage: %s", message);
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}
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void ClDisplayTransientMessage(char* message)
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{
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NSLog(@"DisplayTransientMessage: %s", message);
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}
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-(void) terminate
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{
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// We dispatch this async to get out because this can be invoked
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// on a thread inside common and we don't want to deadlock
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dispatch_async(dispatch_get_main_queue(), ^{
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// This is safe to call even before LiStartConnection
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LiStopConnection();
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});
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}
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-(id) initWithConfig:(StreamConfiguration*)config renderer:(VideoDecoderRenderer*)myRenderer connectionTerminatedCallback:(id<ConTermCallback>)callback
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{
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self = [super init];
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_host = config.hostAddr;
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renderer = myRenderer;
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_callback = callback;
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_streamConfig.width = config.width;
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_streamConfig.height = config.height;
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_streamConfig.fps = config.frameRate;
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_streamConfig.bitrate = config.bitRate;
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_streamConfig.packetSize = 1024;
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memcpy(_streamConfig.remoteInputAesKey, [config.riKey bytes], [config.riKey length]);
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memset(_streamConfig.remoteInputAesIv, 0, 16);
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int riKeyId = htonl(config.riKeyId);
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memcpy(_streamConfig.remoteInputAesIv, &riKeyId, sizeof(riKeyId));
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_drCallbacks.setup = DrSetup;
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_drCallbacks.start = DrStart;
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_drCallbacks.stop = DrStop;
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_drCallbacks.release = DrRelease;
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_drCallbacks.submitDecodeUnit = DrSubmitDecodeUnit;
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_arCallbacks.init = ArInit;
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_arCallbacks.start = ArStart;
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_arCallbacks.stop = ArStop;
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_arCallbacks.release = ArRelease;
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_arCallbacks.decodeAndPlaySample = ArDecodeAndPlaySample;
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_clCallbacks.stageStarting = ClStageStarting;
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_clCallbacks.stageComplete = ClStageComplete;
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_clCallbacks.stageFailed = ClStageFailed;
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_clCallbacks.connectionStarted = ClConnectionStarted;
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_clCallbacks.connectionTerminated = ClConnectionTerminated;
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_clCallbacks.displayMessage = ClDisplayMessage;
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_clCallbacks.displayTransientMessage = ClDisplayTransientMessage;
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return self;
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}
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static OSStatus playbackCallback(void *inRefCon,
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AudioUnitRenderActionFlags *ioActionFlags,
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const AudioTimeStamp *inTimeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList *ioData) {
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// Notes: ioData contains buffers (may be more than one!)
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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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bool ranOutOfData = false;
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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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if (ranOutOfData) {
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ioData->mBuffers[i].mDataByteSize = 0;
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continue;
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}
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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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struct AUDIO_BUFFER_QUEUE_ENTRY *audioEntry = NULL;
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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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audioBufferQueueLength--;
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}
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[audioLock unlock];
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if (audioEntry == NULL) {
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// No data left
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ranOutOfData = true;
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ioData->mBuffers[i].mDataByteSize = thisBufferOffset;
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continue;
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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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audioBufferQueueLength++;
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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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}
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-(void) main
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{
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LiStartConnection(_host, &_streamConfig, &_clCallbacks, &_drCallbacks, &_arCallbacks, NULL, 0);
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}
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@end
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Reference in New Issue
Block a user