我在iOS上做转录应用。所以，我必须将音频记录在缓冲区中，并通过套接字将它们传输到服务器。所以，我使用AudioQueue将音频记录在缓冲区中。
正在本地文件中正确录制音频。对于流式传输，我将音频数据转换为NSData并通过套接字发送。但是，音频质量在服务器中并不好，特别是语音根本不清晰。它在声音的位置包含很多噪音。相同的逻辑在Android中正常工作。所以，服务器端代码工作正常。但是，iOS流式传输转换是一个问题。我使用了两个不同的套接字（SocketRocket / PockSocket）。两个插座中的问题都是一样的。
我在这里附上了我的代码。如果你能帮助我，请告诉我。
ViewController.h
#import <UIKit/UIKit.h>
#import <AudioToolbox/AudioQueue.h>
#import <AudioToolbox/AudioFile.h>
#import <SocketRocket/SocketRocket.h>
#define NUM_BUFFERS 3
#define SAMPLERATE 16000
//Struct defining recording state
typedef struct {
AudioStreamBasicDescription dataFormat;
AudioQueueRef queue;
AudioQueueBufferRef buffers[NUM_BUFFERS];
AudioFileID audioFile;
SInt64 currentPacket;
bool recording;
} RecordState;
//Struct defining playback state
typedef struct {
AudioStreamBasicDescription dataFormat;
AudioQueueRef queue;
AudioQueueBufferRef buffers[NUM_BUFFERS];
AudioFileID audioFile;
SInt64 currentPacket;
bool playing;
} PlayState;
@interface ViewController : UIViewController <SRWebSocketDelegate> {
RecordState recordState;
PlayState playState;
CFURLRef fileURL;
}
@property (nonatomic, strong) SRWebSocket * webSocket;
@property (weak, nonatomic) IBOutlet UITextView *textView;
@end
ViewController.m
#import "ViewController.h"
id thisClass;
//Declare C callback functions
void AudioInputCallback(void * inUserData, // Custom audio metada
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 isNumberPacketDescriptions,
const AudioStreamPacketDescription * inPacketDescs);
void AudioOutputCallback(void * inUserData,
AudioQueueRef outAQ,
AudioQueueBufferRef outBuffer);
@interface ViewController ()
@end
@implementation ViewController
@synthesize webSocket;
@synthesize textView;
// Takes a filled buffer and writes it to disk, "emptying" the buffer
void AudioInputCallback(void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription * inPacketDescs)
{
RecordState * recordState = (RecordState*)inUserData;
if (!recordState->recording)
{
printf("Not recording, returning\n");
}
printf("Writing buffer %lld\n", recordState->currentPacket);
OSStatus status = AudioFileWritePackets(recordState->audioFile,
false,
inBuffer->mAudioDataByteSize,
inPacketDescs,
recordState->currentPacket,
&inNumberPacketDescriptions,
inBuffer->mAudioData);
if (status == 0)
{
recordState->currentPacket += inNumberPacketDescriptions;
NSData * audioData = [NSData dataWithBytes:inBuffer->mAudioData length:inBuffer->mAudioDataByteSize * NUM_BUFFERS];
[thisClass sendAudioToSocketAsData:audioData];
}
AudioQueueEnqueueBuffer(recordState->queue, inBuffer, 0, NULL);
}
// Fills an empty buffer with data and sends it to the speaker
void AudioOutputCallback(void * inUserData,
AudioQueueRef outAQ,
AudioQueueBufferRef outBuffer) {
PlayState * playState = (PlayState *) inUserData;
if(!playState -> playing) {
printf("Not playing, returning\n");
return;
}
printf("Queuing buffer %lld for playback\n", playState -> currentPacket);
AudioStreamPacketDescription * packetDescs;
UInt32 bytesRead;
UInt32 numPackets = SAMPLERATE * NUM_BUFFERS;
OSStatus status;
status = AudioFileReadPackets(playState -> audioFile, false, &bytesRead, packetDescs, playState -> currentPacket, &numPackets, outBuffer -> mAudioData);
if (numPackets) {
outBuffer -> mAudioDataByteSize = bytesRead;
status = AudioQueueEnqueueBuffer(playState -> queue, outBuffer, 0, packetDescs);
playState -> currentPacket += numPackets;
}else {
if (playState -> playing) {
AudioQueueStop(playState -> queue, false);
AudioFileClose(playState -> audioFile);
playState -> playing = false;
}
AudioQueueFreeBuffer(playState -> queue, outBuffer);
}
}
- (void) setupAudioFormat:(AudioStreamBasicDescription *) format {
format -> mSampleRate = SAMPLERATE;
format -> mFormatID = kAudioFormatLinearPCM;
format -> mFramesPerPacket = 1;
format -> mChannelsPerFrame = 1;
format -> mBytesPerFrame = 2;
format -> mBytesPerPacket = 2;
format -> mBitsPerChannel = 16;
format -> mReserved = 0;
format -> mFormatFlags = kLinearPCMFormatFlagIsBigEndian |kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
}
- (void)viewDidLoad {
[super viewDidLoad];
// Do any additional setup after loading the view, typically from a nib.
char path[256];
[self getFilename:path maxLength:sizeof path];
fileURL = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8*)path, strlen(path), false);
// Init state variables
recordState.recording = false;
thisClass = self;
}
- (void) startRecordingInQueue {
[self setupAudioFormat:&recordState.dataFormat];
recordState.currentPacket = 0;
OSStatus status;
status = AudioQueueNewInput(&recordState.dataFormat, AudioInputCallback, &recordState, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &recordState.queue);
if(status == 0) {
//Prime recording buffers with empty data
for (int i=0; i < NUM_BUFFERS; i++) {
AudioQueueAllocateBuffer(recordState.queue, SAMPLERATE, &recordState.buffers[i]);
AudioQueueEnqueueBuffer(recordState.queue, recordState.buffers[i], 0, NULL);
}
status = AudioFileCreateWithURL(fileURL, kAudioFileAIFFType, &recordState.dataFormat, kAudioFileFlags_EraseFile, &recordState.audioFile);
if (status == 0) {
recordState.recording = true;
status = AudioQueueStart(recordState.queue, NULL);
if(status == 0) {
NSLog(@"-----------Recording--------------");
NSLog(@"File URL : %@", fileURL);
}
}
}
if (status != 0) {
[self stopRecordingInQueue];
}
}
- (void) stopRecordingInQueue {
recordState.recording = false;
AudioQueueStop(recordState.queue, true);
for (int i=0; i < NUM_BUFFERS; i++) {
AudioQueueFreeBuffer(recordState.queue, recordState.buffers[i]);
}
AudioQueueDispose(recordState.queue, true);
AudioFileClose(recordState.audioFile);
NSLog(@"---Idle------");
NSLog(@"File URL : %@", fileURL);
}
- (void) startPlaybackInQueue {
playState.currentPacket = 0;
[self setupAudioFormat:&playState.dataFormat];
OSStatus status;
status = AudioFileOpenURL(fileURL, kAudioFileReadPermission, kAudioFileAIFFType, &playState.audioFile);
if (status == 0) {
status = AudioQueueNewOutput(&playState.dataFormat, AudioOutputCallback, &playState, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &playState.queue);
if( status == 0) {
//Allocate and prime playback buffers
playState.playing = true;
for (int i=0; i < NUM_BUFFERS && playState.playing; i++) {
AudioQueueAllocateBuffer(playState.queue, SAMPLERATE, &playState.buffers[i]);
AudioOutputCallback(&playState, playState.queue, playState.buffers[i]);
}
status = AudioQueueStart(playState.queue, NULL);
if (status == 0) {
NSLog(@"-------Playing Audio---------");
}
}
}
if (status != 0) {
[self stopPlaybackInQueue];
NSLog(@"---Playing Audio Failed ------");
}
}
- (void) stopPlaybackInQueue {
playState.playing = false;
for (int i=0; i < NUM_BUFFERS; i++) {
AudioQueueFreeBuffer(playState.queue, playState.buffers[i]);
}
AudioQueueDispose(playState.queue, true);
AudioFileClose(playState.audioFile);
}
- (IBAction)startRecordingAudio:(id)sender {
NSLog(@"starting recording tapped");
[self startRecordingInQueue];
}
- (IBAction)stopRecordingAudio:(id)sender {
NSLog(@"stop recording tapped");
[self stopRecordingInQueue];
}
- (IBAction)startPlayingAudio:(id)sender {
NSLog(@"start playing audio tapped");
[self startPlaybackInQueue];
}
- (IBAction)stopPlayingAudio:(id)sender {
NSLog(@"stop playing audio tapped");
[self stopPlaybackInQueue];
}
- (BOOL) getFilename:(char *) buffer maxLength:(int) maxBufferLength {
NSArray * paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString * docDir = [paths objectAtIndex:0];
NSString * file = [docDir stringByAppendingString:@"recording.aif"];
return [file getCString:buffer maxLength:maxBufferLength encoding:NSUTF8StringEncoding];
}
- (void) sendAudioToSocketAsData:(NSData *) audioData {
[self.webSocket send:audioData];
}
- (IBAction)connectToSocketTapped:(id)sender {
[self startStreaming];
}
- (void) startStreaming {
[self connectToSocket];
}
- (void) connectToSocket {
//Socket Connection Intiliazation
// create the NSURLRequest that will be sent as the handshake
NSURLRequest *request = [NSURLRequest requestWithURL:[NSURL URLWithString:@"${url}"]];
// create the socket and assign delegate
self.webSocket = [[SRWebSocket alloc] initWithURLRequest:request];
self.webSocket.delegate = self;
// open socket
[self.webSocket open];
}
///--------------------------------------
#pragma mark - SRWebSocketDelegate
///--------------------------------------
- (void)webSocketDidOpen:(SRWebSocket *)webSocket;
{
NSLog(@"Websocket Connected");
}
- (void) webSocket:(SRWebSocket *)webSocket didFailWithError:(NSError *)error {
NSLog(@":( Websocket Failed With Error %@", error);
self.webSocket = nil;
}
- (void) webSocket:(SRWebSocket *)webSocket didReceiveMessage:(id)message {
NSLog(@"Received \"%@\"", message);
textView.text = message;
}
- (void)webSocket:(SRWebSocket *)webSocket didCloseWithCode:(NSInteger)code reason:(NSString *)reason wasClean:(BOOL)wasClean;
{
NSLog(@"WebSocket closed");
self.webSocket = nil;
}
- (void)webSocket:(SRWebSocket *)webSocket didReceivePong:(NSData *)pongPayload;
{
NSLog(@"WebSocket received pong");
}
- (void)didReceiveMemoryWarning {
[super didReceiveMemoryWarning];
// Dispose of any resources that can be recreated.
}
先谢谢
答案 0 :(得分：1)
我做到了。设置的音频格式导致了问题。我通过检查服务器端文档正确设置了音频。 Big-Endian造成了问题。如果将其指定为big-endian，则为big endian。如果你没有指定它，那么它就是little-endian。我需要小端。
- (void) setupAudioFormat:(AudioStreamBasicDescription *) format {
format -> mSampleRate = 16000.0; //
format -> mFormatID = kAudioFormatLinearPCM; //
format -> mFramesPerPacket = 1;
format -> mChannelsPerFrame = 1; //
format -> mBytesPerFrame = 2;
format -> mBytesPerPacket = 2;
format -> mBitsPerChannel = 16; //
// format -> mReserved = 0;
format -> mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
}