私の調査に基づくと、人々はそれが不一致/間違ったフォーマットに基づいていると言い続けています。しかし、私は入力と出力の両方にlPCMフォーマットを使用しています。私が得ている結果はただのノイズです..（ホワイトノイズのように）
コード全体を貼り付けることにしました。おそらくそれが役立つでしょう：
#import "AppDelegate.h"
#import "ViewController.h"
@implementation AppDelegate
@synthesize window = _window;
@synthesize viewController = _viewController;
- (BOOL)application:(UIApplication *)application didFinishLaunchingWithOptions:(NSDictionary *)launchOptions
{
self.window = [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]];
// Override point for customization after application launch.
self.viewController = [[ViewController alloc] initWithNibName:@"ViewController" bundle:nil];
self.window.rootViewController = self.viewController;
[self.window makeKeyAndVisible];
// Insert code here to initialize your application
player = [[Player alloc] init];
[self setupReader];
[self setupQueue];
// initialize reader in a new thread
internalThread =[[NSThread alloc]
initWithTarget:self
selector:@selector(readPackets)
object:nil];
[internalThread start];
// start the queue. this function returns immedatly and begins
// invoking the callback, as needed, asynchronously.
//CheckError(AudioQueueStart(queue, NULL), "AudioQueueStart failed");
// and wait
printf("Playing...\n");
do
{
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.25, false);
} while (!player.isDone /*|| gIsRunning*/);
// isDone represents the state of the Audio File enqueuing. This does not mean the
// Audio Queue is actually done playing yet. Since we have 3 half-second buffers in-flight
// run for continue to run for a short additional time so they can be processed
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 2, false);
// end playback
player.isDone = true;
CheckError(AudioQueueStop(queue, TRUE), "AudioQueueStop failed");
cleanup:
AudioQueueDispose(queue, TRUE);
AudioFileClose(player.playbackFile);
return YES;
}
- (void) setupReader
{
NSURL *assetURL = [NSURL URLWithString:@"ipod-library://item/item.m4a?id=1053020204400037178"]; // from ilham's ipod
AVURLAsset *songAsset = [AVURLAsset URLAssetWithURL:assetURL options:nil];
// from AVAssetReader Class Reference:
// AVAssetReader is not intended for use with real-time sources,
// and its performance is not guaranteed for real-time operations.
NSError * error = nil;
AVAssetReader* reader = [[AVAssetReader alloc] initWithAsset:songAsset error:&error];
AVAssetTrack* track = [songAsset.tracks objectAtIndex:0];
readerOutput = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:track
outputSettings:nil];
// AVAssetReaderOutput* readerOutput = [[AVAssetReaderAudioMixOutput alloc] initWithAudioTracks:songAsset.tracks audioSettings:nil];
[reader addOutput:readerOutput];
[reader startReading];
}
- (void) setupQueue
{
// get the audio data format from the file
// we know that it is PCM.. since it's converted
AudioStreamBasicDescription dataFormat;
dataFormat.mSampleRate = 44100.0;
dataFormat.mFormatID = kAudioFormatLinearPCM;
dataFormat.mFormatFlags = kAudioFormatFlagIsBigEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
dataFormat.mBytesPerPacket = 4;
dataFormat.mFramesPerPacket = 1;
dataFormat.mBytesPerFrame = 4;
dataFormat.mChannelsPerFrame = 2;
dataFormat.mBitsPerChannel = 16;
// create a output (playback) queue
CheckError(AudioQueueNewOutput(&dataFormat, // ASBD
MyAQOutputCallback, // Callback
(__bridge void *)self, // user data
NULL, // run loop
NULL, // run loop mode
0, // flags (always 0)
&queue), // output: reference to AudioQueue object
"AudioQueueNewOutput failed");
// adjust buffer size to represent about a half second (0.5) of audio based on this format
CalculateBytesForTime(dataFormat, 0.5, &bufferByteSize, &player->numPacketsToRead);
// check if we are dealing with a VBR file. ASBDs for VBR files always have
// mBytesPerPacket and mFramesPerPacket as 0 since they can fluctuate at any time.
// If we are dealing with a VBR file, we allocate memory to hold the packet descriptions
bool isFormatVBR = (dataFormat.mBytesPerPacket == 0 || dataFormat.mFramesPerPacket == 0);
if (isFormatVBR)
player.packetDescs = (AudioStreamPacketDescription*)malloc(sizeof(AudioStreamPacketDescription) * player.numPacketsToRead);
else
player.packetDescs = NULL; // we don't provide packet descriptions for constant bit rate formats (like linear PCM)
// get magic cookie from file and set on queue
MyCopyEncoderCookieToQueue(player.playbackFile, queue);
// allocate the buffers and prime the queue with some data before starting
player.isDone = false;
player.packetPosition = 0;
int i;
for (i = 0; i < kNumberPlaybackBuffers; ++i)
{
CheckError(AudioQueueAllocateBuffer(queue, bufferByteSize, &audioQueueBuffers[i]), "AudioQueueAllocateBuffer failed");
// EOF (the entire file's contents fit in the buffers)
if (player.isDone)
break;
}
}
-(void)readPackets
{
// initialize a mutex and condition so that we can block on buffers in use.
pthread_mutex_init(&queueBuffersMutex, NULL);
pthread_cond_init(&queueBufferReadyCondition, NULL);
state = AS_BUFFERING;
while ((sample = [readerOutput copyNextSampleBuffer])) {
AudioBufferList audioBufferList;
CMBlockBufferRef CMBuffer = CMSampleBufferGetDataBuffer( sample );
CheckError(CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(
sample,
NULL,
&audioBufferList,
sizeof(audioBufferList),
NULL,
NULL,
kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment,
&CMBuffer
),
"could not read samples");
AudioBuffer audioBuffer = audioBufferList.mBuffers[0];
UInt32 inNumberBytes = audioBuffer.mDataByteSize;
size_t incomingDataOffset = 0;
while (inNumberBytes) {
size_t bufSpaceRemaining;
bufSpaceRemaining = bufferByteSize - bytesFilled;
@synchronized(self)
{
bufSpaceRemaining = bufferByteSize - bytesFilled;
size_t copySize;
if (bufSpaceRemaining < inNumberBytes)
{
copySize = bufSpaceRemaining;
}
else
{
copySize = inNumberBytes;
}
// copy data to the audio queue buffer
AudioQueueBufferRef fillBuf = audioQueueBuffers[fillBufferIndex];
memcpy((char*)fillBuf->mAudioData + bytesFilled, (const char*)(audioBuffer.mData + incomingDataOffset), copySize);
// keep track of bytes filled
bytesFilled +=copySize;
incomingDataOffset +=copySize;
inNumberBytes -=copySize;
}
// if the space remaining in the buffer is not enough for this packet, then enqueue the buffer.
if (bufSpaceRemaining < inNumberBytes + bytesFilled)
{
[self enqueueBuffer];
}
}
}
}
-(void)enqueueBuffer
{
@synchronized(self)
{
inuse[fillBufferIndex] = true; // set in use flag
buffersUsed++;
// enqueue buffer
AudioQueueBufferRef fillBuf = audioQueueBuffers[fillBufferIndex];
NSLog(@"we are now enqueing buffer %d",fillBufferIndex);
fillBuf->mAudioDataByteSize = bytesFilled;
err = AudioQueueEnqueueBuffer(queue, fillBuf, 0, NULL);
if (err)
{
NSLog(@"could not enqueue queue with buffer");
return;
}
if (state == AS_BUFFERING)
{
//
// Fill all the buffers before starting. This ensures that the
// AudioFileStream stays a small amount ahead of the AudioQueue to
// avoid an audio glitch playing streaming files on iPhone SDKs < 3.0
//
if (buffersUsed == kNumberPlaybackBuffers - 1)
{
err = AudioQueueStart(queue, NULL);
if (err)
{
NSLog(@"couldn't start queue");
return;
}
state = AS_PLAYING;
}
}
// go to next buffer
if (++fillBufferIndex >= kNumberPlaybackBuffers) fillBufferIndex = 0;
bytesFilled = 0; // reset bytes filled
}
// wait until next buffer is not in use
pthread_mutex_lock(&queueBuffersMutex);
while (inuse[fillBufferIndex])
{
pthread_cond_wait(&queueBufferReadyCondition, &queueBuffersMutex);
}
pthread_mutex_unlock(&queueBuffersMutex);
}
#pragma mark - utility functions -
// generic error handler - if err is nonzero, prints error message and exits program.
static void CheckError(OSStatus error, const char *operation)
{
if (error == noErr) return;
char str[20];
// see if it appears to be a 4-char-code
*(UInt32 *)(str + 1) = CFSwapInt32HostToBig(error);
if (isprint(str[1]) && isprint(str[2]) && isprint(str[3]) && isprint(str[4])) {
str[0] = str[5] = '\'';
str[6] = '\0';
} else
// no, format it as an integer
sprintf(str, "%d", (int)error);
fprintf(stderr, "Error: %s (%s)\n", operation, str);
exit(1);
}
// we only use time here as a guideline
// we're really trying to get somewhere between 16K and 64K buffers, but not allocate too much if we don't need it/*
void CalculateBytesForTime(AudioStreamBasicDescription inDesc, Float64 inSeconds, UInt32 *outBufferSize, UInt32 *outNumPackets)
{
// we need to calculate how many packets we read at a time, and how big a buffer we need.
// we base this on the size of the packets in the file and an approximate duration for each buffer.
//
// first check to see what the max size of a packet is, if it is bigger than our default
// allocation size, that needs to become larger
// we don't have access to file packet size, so we just default it to maxBufferSize
UInt32 maxPacketSize = 0x10000;
static const int maxBufferSize = 0x10000; // limit size to 64K
static const int minBufferSize = 0x4000; // limit size to 16K
if (inDesc.mFramesPerPacket) {
Float64 numPacketsForTime = inDesc.mSampleRate / inDesc.mFramesPerPacket * inSeconds;
*outBufferSize = numPacketsForTime * maxPacketSize;
} else {
// if frames per packet is zero, then the codec has no predictable packet == time
// so we can't tailor this (we don't know how many Packets represent a time period
// we'll just return a default buffer size
*outBufferSize = maxBufferSize > maxPacketSize ? maxBufferSize : maxPacketSize;
}
// we're going to limit our size to our default
if (*outBufferSize > maxBufferSize && *outBufferSize > maxPacketSize)
*outBufferSize = maxBufferSize;
else {
// also make sure we're not too small - we don't want to go the disk for too small chunks
if (*outBufferSize < minBufferSize)
*outBufferSize = minBufferSize;
}
*outNumPackets = *outBufferSize / maxPacketSize;
}
// many encoded formats require a 'magic cookie'. if the file has a cookie we get it
// and configure the queue with it
static void MyCopyEncoderCookieToQueue(AudioFileID theFile, AudioQueueRef queue ) {
UInt32 propertySize;
OSStatus result = AudioFileGetPropertyInfo (theFile, kAudioFilePropertyMagicCookieData, &propertySize, NULL);
if (result == noErr && propertySize > 0)
{
Byte* magicCookie = (UInt8*)malloc(sizeof(UInt8) * propertySize);
CheckError(AudioFileGetProperty (theFile, kAudioFilePropertyMagicCookieData, &propertySize, magicCookie), "get cookie from file failed");
CheckError(AudioQueueSetProperty(queue, kAudioQueueProperty_MagicCookie, magicCookie, propertySize), "set cookie on queue failed");
free(magicCookie);
}
}
#pragma mark - audio queue -
static void MyAQOutputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inCompleteAQBuffer)
{
AppDelegate *appDelegate = (__bridge AppDelegate *) inUserData;
[appDelegate myCallback:inUserData
inAudioQueue:inAQ
audioQueueBufferRef:inCompleteAQBuffer];
}
- (void)myCallback:(void *)userData
inAudioQueue:(AudioQueueRef)inAQ
audioQueueBufferRef:(AudioQueueBufferRef)inCompleteAQBuffer
{
unsigned int bufIndex = -1;
for (unsigned int i = 0; i < kNumberPlaybackBuffers; ++i)
{
if (inCompleteAQBuffer == audioQueueBuffers[i])
{
bufIndex = i;
break;
}
}
if (bufIndex == -1)
{
NSLog(@"something went wrong at queue callback");
return;
}
// signal waiting thread that the buffer is free.
pthread_mutex_lock(&queueBuffersMutex);
NSLog(@"signalling that buffer %d is free",bufIndex);
inuse[bufIndex] = false;
buffersUsed--;
pthread_cond_signal(&queueBufferReadyCondition);
pthread_mutex_unlock(&queueBuffersMutex);
}
@end
更新：以下の btomwの回答は、問題を大幅に解決しました。しかし、私はこれの根底に行きたいと思います（私のようなほとんどの初心者開発者、そして彼が最初に始めたときのbtomwでさえ、通常、パラメーター、フォーマットなどで暗闇の中で撮影します-例についてはここを参照してください-）。
AVURLAssetのパラメーターとしてnulを指定した理由*songAsset= [AVURLAsset URLAssetWithURL：assetURL options：audioReadSettings];
ドキュメントと試行錯誤によると、lPCM以外のフォーマットは完全に拒否されることに気付いたからです。つまり、AVAseetReaderまたは変換を使用すると、結果は常にlPCM ..であるため、デフォルトの形式はとにかくlPCMであると思ったので、nullのままにしました..しかし、私は間違っていたと思います。
これの奇妙な部分（私が間違っている場合は誰かを訂正してください）は、私が述べたように..元のファイルが.mp3であると仮定し、それを再生する（またはネットワーク経由でパケットを送信するなど）ことを意図したことですmp3として..そして私はmp3ABSDを提供しました..アセットリーダーがクラッシュします！それで、元の形式で送信したい場合は、nullを指定するだけですか？これに関する明らかな問題は、反対側で受け取ったABSDが何であるかを理解する方法がないということです。
アップデート2：githubからコードをダウンロードできます。