Question

我有以下代码打开一个AudioQueue来播放16位pcm @ 44,100hz。它有一个非常奇怪的怪癖，一旦初始缓冲区被填满，它就会很快播放，然后在等待更多字节通过网络时变得“波动”。

所以要么我搞砸了将数据子范围复制到缓冲区的代码，要么我已经告诉AudioQueue比数据通过网络更快地播放。

有人有什么想法吗？我已经被困了几天了。

//
// Created by Benjamin St Pierre on 15-01-02.
// Copyright (c) 2015 Lightning Strike Solutions. All rights reserved.
//

#import <MacTypes.h>
#import "MediaPlayer.h"


@implementation MediaPlayer


@synthesize sampleQueue;


void OutputBufferCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
    //Cast userData to MediaPlayer Objective-C class instance
    MediaPlayer *mediaPlayer = (__bridge MediaPlayer *) inUserData;
    // Fill buffer.
    [mediaPlayer fillAudioBuffer:inBuffer];
    // Re-enqueue buffer.
    OSStatus err = AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
    if (err != noErr)
        NSLog(@"AudioQueueEnqueueBuffer() error %d", (int) err);
}

- (void)fillAudioBuffer:(AudioQueueBufferRef)inBuffer {
    if (self.currentAudioPiece == nil || self.currentAudioPiece.duration >= self.currentAudioPieceIndex) {
        //grab latest sample from sample queue
        self.currentAudioPiece = sampleQueue.dequeue;
        self.currentAudioPieceIndex = 0;
    }

    //Check for empty sample queue
    if (self.currentAudioPiece == nil) {
        NSLog(@"Empty sample queue");
        memset(inBuffer->mAudioData, 0, kBufferByteSize);
        return;
    }

    UInt32 bytesToRead = inBuffer->mAudioDataBytesCapacity;

    while (bytesToRead > 0) {
        UInt32 maxBytesFromCurrentPiece = self.currentAudioPiece.audioData.length - self.currentAudioPieceIndex;
        //Take the min of what the current piece can provide OR what is needed to be read
        UInt32 bytesToReadNow = MIN(maxBytesFromCurrentPiece, bytesToRead);

        NSData *subRange = [self.currentAudioPiece.audioData subdataWithRange:NSMakeRange(self.currentAudioPieceIndex, bytesToReadNow)];
        //Copy what you can before continuing loop
        memcpy(inBuffer->mAudioData, subRange.bytes, subRange.length);
        bytesToRead -= bytesToReadNow;

        if (bytesToReadNow == maxBytesFromCurrentPiece) {
            @synchronized (sampleQueue) {
                self.currentAudioPiece = self.sampleQueue.dequeue;
                self.currentAudioPieceIndex = 0;
            }
        } else {
            self.currentAudioPieceIndex += bytesToReadNow;
        }
    }
    inBuffer->mAudioDataByteSize = kBufferByteSize;
}

- (void)startMediaPlayer {
    AudioStreamBasicDescription streamFormat;
    streamFormat.mFormatID = kAudioFormatLinearPCM;
    streamFormat.mSampleRate = 44100.0;
    streamFormat.mChannelsPerFrame = 2;
    streamFormat.mBytesPerFrame = 4;
    streamFormat.mFramesPerPacket = 1;
    streamFormat.mBytesPerPacket = 4;
    streamFormat.mBitsPerChannel = 16;
    streamFormat.mReserved = 0;
    streamFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;

    // New input queue
    OSStatus err = AudioQueueNewOutput(&streamFormat, OutputBufferCallback, (__bridge void *) self, nil, nil, 0, &outputQueue);
    if (err != noErr) {
        NSLog(@"AudioQueueNewOutput() error: %d", (int) err);
    }

    int i;
    // Enqueue buffers
    AudioQueueBufferRef buffer;
    for (i = 0; i < kNumberBuffers; i++) {
        err = AudioQueueAllocateBuffer(outputQueue, kBufferByteSize, &buffer);
        memset(buffer->mAudioData, 0, kBufferByteSize);
        buffer->mAudioDataByteSize = kBufferByteSize;
        if (err == noErr) {
            err = AudioQueueEnqueueBuffer(outputQueue, buffer, 0, nil);
            if (err != noErr) NSLog(@"AudioQueueEnqueueBuffer() error: %d", (int) err);
        } else {
            NSLog(@"AudioQueueAllocateBuffer() error: %d", (int) err);
            return;
        }
    }

    // Start queue
    err = AudioQueueStart(outputQueue, nil);
    if (err != noErr) NSLog(@"AudioQueueStart() error: %d", (int) err);
}

@end

Answer 1

我打算在这里唠叨，并说你的播放不稳定，因为你没有推进数据的写指针。我不知道Objective-C是否足以告诉你这个语法是否正确，但我认为你需要添加：

while (bytesToRead > 0) {
    ....
    memcpy(inBuffer->mAudioData, subRange.bytes, subRange.length);
    bytesToRead -= bytesToReadNow;
    inBuffer->mAudioData += bytesReadNow; // move the write pointer
    ...
}

使用AudioQueue进行Choppy音频播放

1 个答案: