I am using https://github.com/tumtumtum/StreamingKit
for streaming a live url. works perfectly fine. I want to add recording/save of audio feature to my app. Does anyone know if this library can do this?
If not, are there any alternatives? NOTE that I need to record LIVE streaming audio, not local file / static url.
The page shows that you can Intercept PCM data just before its played:
[audioPlayer appendFrameFilterWithName:#"MyCustomFilter" block:^(UInt32 channelsPerFrame, UInt32 bytesPerFrame, UInt32 frameCount, void* frames)
{
...
}];
However, I am not sure how to convert this into actual recording / mp3 file or even intercept the actual data from this?
You can do something like this, although StreamingKit seems a little bit secretive about the format of the samples it gives you. What's the sample rate? Floats or ints? I suppose you can guess from the sample size. This example assumes 16bit ints.
NSURL *dstUrl = [[NSURL fileURLWithPath:NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES)[0] ] URLByAppendingPathComponent:#"output.m4a"];
NSLog(#"write to %#", dstUrl);
__block AVAudioFile *audioFile = nil;
[audioPlayer appendFrameFilterWithName:#"MyCustomFilter" block:^(UInt32 channelsPerFrame, UInt32 bytesPerFrame, UInt32 frameCount, void* frames)
{
NSError *error;
// what's the sample rate? StreamingKit doesn't seem to tell us
double sampleRate = 44100;
if (!audioFile) {
NSDictionary *settings =
#{
AVFormatIDKey : #(kAudioFormatMPEG4AAC),
AVSampleRateKey : #(sampleRate),
AVNumberOfChannelsKey : #(channelsPerFrame),
};
// need commonFormat?
audioFile = [[AVAudioFile alloc] initForWriting:dstUrl settings:settings commonFormat:AVAudioPCMFormatInt16 interleaved:YES error:&error];
if (!audioFile) {
// error
}
}
AVAudioFormat *format = [[AVAudioFormat alloc] initWithCommonFormat:AVAudioPCMFormatInt16 sampleRate:sampleRate channels:channelsPerFrame interleaved:YES];
AVAudioPCMBuffer *buffer = [[AVAudioPCMBuffer alloc] initWithPCMFormat:format frameCapacity:frameCount];
buffer.frameLength = frameCount;
memmove(buffer.int16ChannelData[0], frames, frameCount*bytesPerFrame);
if (![audioFile writeFromBuffer:buffer error:&error]) {
NSLog(#"write error: %#", error);
}
}];
[self.audioPlayer performSelector:#selector(removeFrameFilterWithName:) withObject:#"MyCustomFilter" afterDelay:10];
I am playing HLS streams using AVPlayer. And I also need to record these streams as user presses record button.
The approach I am using is to record audio and video separately then at the end merge these file to make the final video. And It is successful with remote mp4 files.
But now for the HLS (.m3u8) files I am able to record the video using AVAssetWriter but having problems with audio recording.
I am using MTAudioProccessingTap to process the raw audio data and write it to a file. I followed this article. I am able to record remote mp4 audio but its not working with HLS streams.
Initially I wasn't able to extract the audio tracks from the stream using AVAssetTrack *audioTrack = [asset tracksWithMediaType:AVMediaTypeAudio][0];
But I was able to extract the audioTracks using KVO to initialize the MTAudioProcessingTap.
-(void)observeValueForKeyPath:(NSString *)keyPath ofObject:(id)object change:(NSDictionary *)change context:(void *)context{
AVPlayer *player = (AVPlayer*) object;
if (player.status == AVPlayerStatusReadyToPlay)
{
NSLog(#"Ready to play");
self.previousAudioTrackID = 0;
__weak typeof (self) weakself = self;
timeObserverForTrack = [player addPeriodicTimeObserverForInterval:CMTimeMakeWithSeconds(1, 100) queue:nil usingBlock:^(CMTime time)
{
#try {
for(AVPlayerItemTrack* track in [weakself.avPlayer.currentItem tracks]) {
if([track.assetTrack.mediaType isEqualToString:AVMediaTypeAudio])
weakself.currentAudioPlayerItemTrack = track;
}
AVAssetTrack* audioAssetTrack = weakself.currentAudioPlayerItemTrack.assetTrack;
weakself.currentAudioTrackID = audioAssetTrack.trackID;
if(weakself.previousAudioTrackID != weakself.currentAudioTrackID) {
NSLog(#":::::::::::::::::::::::::: Audio track changed : %d",weakself.currentAudioTrackID);
weakself.previousAudioTrackID = weakself.currentAudioTrackID;
weakself.audioTrack = audioAssetTrack;
/// Use this audio track to initialize MTAudioProcessingTap
}
}
#catch (NSException *exception) {
NSLog(#"Exception Trap ::::: Audio tracks not found!");
}
}];
}
}
I am also keeping track of trackID to check if track is changed.
This is how I initialize the MTAudioProcessingTap.
-(void)beginRecordingAudioFromTrack:(AVAssetTrack *)audioTrack{
// Configure an MTAudioProcessingTap to handle things.
MTAudioProcessingTapRef tap;
MTAudioProcessingTapCallbacks callbacks;
callbacks.version = kMTAudioProcessingTapCallbacksVersion_0;
callbacks.clientInfo = (__bridge void *)(self);
callbacks.init = init;
callbacks.prepare = prepare;
callbacks.process = process;
callbacks.unprepare = unprepare;
callbacks.finalize = finalize;
OSStatus err = MTAudioProcessingTapCreate(
kCFAllocatorDefault,
&callbacks,
kMTAudioProcessingTapCreationFlag_PostEffects,
&tap
);
if(err) {
NSLog(#"Unable to create the Audio Processing Tap %d", (int)err);
NSError *error = [NSError errorWithDomain:NSOSStatusErrorDomain
code:err
userInfo:nil];
NSLog(#"Error: %#", [error description]);;
return;
}
// Create an AudioMix and assign it to our currently playing "item", which
// is just the stream itself.
AVMutableAudioMix *audioMix = [AVMutableAudioMix audioMix];
AVMutableAudioMixInputParameters *inputParams = [AVMutableAudioMixInputParameters
audioMixInputParametersWithTrack:audioTrack];
inputParams.audioTapProcessor = tap;
audioMix.inputParameters = #[inputParams];
_audioPlayer.currentItem.audioMix = audioMix;
}
But Now with this audio track MTAudioProcessingTap callbacks "Prepare" and "Process" are never called.
Is the problem with the audioTrack I am getting through KVO?
Now I would really appreciate if some one can help me with this. Or can tell am I using the write approach to record HLS Streams?
I Found solution for this and using it in my app. Wanted to post it earlier but didn't get the time.
So to play with HLS you should have some knowledge what they are exactly. For that please see it here on Apple Website.
HLS Apple
Here are the steps I am following.
1. First get the m3u8 and parse it.
You can parse it using this helpful kit M3U8Kit.
Using this kit you can get the M3U8MediaPlaylist or M3U8MasterPlaylist(if it is a master playlist)
if you get the master playlist you can also parse it to get M3U8MediaPlaylist
(void) parseM3u8
{
NSString *plainString = [self.url m3u8PlanString];
BOOL isMasterPlaylist = [plainString isMasterPlaylist];
NSError *error;
NSURL *baseURL;
if(isMasterPlaylist)
{
M3U8MasterPlaylist *masterList = [[M3U8MasterPlaylist alloc] initWithContentOfURL:self.url error:&error];
self.masterPlaylist = masterList;
M3U8ExtXStreamInfList *xStreamInfList = masterList.xStreamList;
M3U8ExtXStreamInf *StreamInfo = [xStreamInfList extXStreamInfAtIndex:0];
NSString *URI = StreamInfo.URI;
NSRange range = [URI rangeOfString:#"dailymotion.com"];
NSString *baseURLString = [URI substringToIndex:(range.location+range.length)];
baseURL = [NSURL URLWithString:baseURLString];
plainString = [[NSURL URLWithString:URI] m3u8PlanString];
}
M3U8MediaPlaylist *mediaPlaylist = [[M3U8MediaPlaylist alloc] initWithContent:plainString baseURL:baseURL];
self.mediaPlaylist = mediaPlaylist;
M3U8SegmentInfoList *segmentInfoList = mediaPlaylist.segmentList;
NSMutableArray *segmentUrls = [[NSMutableArray alloc] init];
for (int i = 0; i < segmentInfoList.count; i++)
{
M3U8SegmentInfo *segmentInfo = [segmentInfoList segmentInfoAtIndex:i];
NSString *segmentURI = segmentInfo.URI;
NSURL *mediaURL = [baseURL URLByAppendingPathComponent:segmentURI];
[segmentUrls addObject:mediaURL];
if(!self.segmentDuration)
self.segmentDuration = segmentInfo.duration;
}
self.segmentFilesURLs = segmentUrls;
}
You can see that you will get the links to the .ts files from the m3u8 parse it.
Now download all the .ts file into a local folder.
Merge these .ts files in to one mp4 file and Export.
You can do that using this wonderful C library
TS2MP4
and then you can delete the .ts files or keep them if you need them.
This is not good approach what you can do is to Parse M3U8 link .Then try to download segment files (.ts) . If you can get these file you can merge them to generate mp4 file.
i'm using OpenAL to play background music and sound effects in a game, the code to play background music is this;
- (void) playSoundBack
{
alGenSources(1, &sourceIDBack);
NSString *audioFilePath = [[NSBundle mainBundle] pathForResource:#"music" ofType:#"wav"];
NSURL *audioFileURL = [NSURL fileURLWithPath:audioFilePath];
AudioFileID afid;
OSStatus openAudioFileResult = AudioFileOpenURL((__bridge CFURLRef)audioFileURL, kAudioFileReadPermission, 0, &afid);
if (0 != openAudioFileResult)
{
NSLog(#"An error occurred when attempting to open the audio file %#: %ld", audioFilePath, openAudioFileResult);
return;
}
UInt64 audioDataByteCount = 0;
UInt32 propertySize = sizeof(audioDataByteCount);
OSStatus getSizeResult = AudioFileGetProperty(afid, kAudioFilePropertyAudioDataByteCount, &propertySize, &audioDataByteCount);
if (0 != getSizeResult)
{
NSLog(#"An error occurred when attempting to determine the size of audio file %#: %ld", audioFilePath, getSizeResult);
}
UInt32 bytesRead = (UInt32)audioDataByteCount;
void *audioData = malloc(bytesRead);
OSStatus readBytesResult = AudioFileReadBytes(afid, false, 0, &bytesRead, audioData);
if (0 != readBytesResult)
{
NSLog(#"An error occurred when attempting to read data from audio file %#: %ld", audioFilePath, readBytesResult);
}
AudioFileClose(afid);
ALuint outputBuffer;
alGenBuffers(1, &outputBuffer);
alBufferData(outputBuffer, AL_FORMAT_STEREO16, audioData, bytesRead, 44100);
if (audioData)
{
free(audioData);
audioData = NULL;
}
alSourcef(sourceIDBack, AL_GAIN, 0.1f);
alSourcei(sourceIDBack, AL_BUFFER, outputBuffer);
alSourcei(sourceIDBack, AL_LOOPING, AL_TRUE);
alSourcePlay(sourceIDBack);
}
With another similar block of code i play some effects during the game. All the sound are played but i have a problem with volumes... The background music volume change and it seems not an absolute value... When the other effects are playing the background music volume is low as i have set in with AL_GAIN,but when no other effect is playing the background music sounds too loud (or at least this is my sensation...). Why the music has this behaviour? The volume setting like 0.1 are relative? It is possible to set a fixed and universal value?
I writing an iOS app that needs to identify periods of silence within an mp3 file downloaded from the internet.
Downloading and playing via AVAudioPlayer is no problem but I can't figure out how to get access to the actual audio frame data within the mp3 so that I can detect sound levels in order to detect silences. I've tried :-
NSURL *fileUrl = [[NSURL alloc] initFileURLWithPath:[self filename]];
ExtAudioFileRef eaf;
OSStatus err = ExtAudioFileOpenURL((CFURLRef)CFBridgingRetain(fileUrl), &eaf);
if (noErr != err)
{
/* handle error */
exit(-1);
}
AudioStreamBasicDescription format;
format.mSampleRate = 44100;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kAudioFormatFlagsNativeEndian;
format.mBitsPerChannel = 16;
format.mChannelsPerFrame = 2;
format.mBytesPerFrame = format.mChannelsPerFrame * 2;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mFramesPerPacket * format.mBytesPerFrame;
err = ExtAudioFileSetProperty(eaf, kExtAudioFileProperty_ClientDataFormat, sizeof(format), &format);
if (noErr != err)
{
/* handle error */
NSLog(#"Error: %d", err);
exit(-1);
}
But it fails on the set property with error 1718449215, so I'm assuming that it won't allow me to convert from MP3 to PCM. I want to be able to preprocess the files so that I know where the silences are going to occur before playback by the user.
Any suggestions would be greatly appreciated.
Thanks.
You can use the AVAssetReader and AVAssetWriter APIs to convert an mp3 file into raw PCM (or .wav) data.
I am currently working on an application as part of my Bachelor in Computer Science. The application will correlate data from the iPhone hardware (accelerometer, gps) and music that is being played.
The project is still in its infancy, having worked on it for only 2 months.
The moment that I am right now, and where I need help, is reading PCM samples from songs from the itunes library, and playing them back using and audio unit.
Currently the implementation I would like working does the following: chooses a random song from iTunes, and reads samples from it when required, and stores in a buffer, lets call it sampleBuffer. Later on in the consumer model the audio unit (which has a mixer and a remoteIO output) has a callback where I simply copy the required number of samples from sampleBuffer into the buffer specified in the callback. What i then hear through the speakers is something not quite what i expect; I can recognize that it is playing the song however it seems that it is incorrectly decoded and it has a lot of noise! I attached an image which shows the first ~half a second (24576 samples # 44.1kHz), and this does not resemble a normall looking output.
Before I get into the listing I have checked that the file is not corrupted, similarily I have written test cases for the buffer (so I know the buffer does not alter the samples), and although this might not be the best way to do it (some would argue to go the audio queue route), I want to perform various manipulations on the samples aswell as changing the song before it is finished, rearranging what song is played, etc. Furthermore, maybe there are some incorrect settings in the audio unit, however, the graph that displays the samples (which shows the samples are decoded incorrectly) is taken straight from the buffer, thus I am only looking now to solve why the reading from the disk and decoding does not work correctly. Right now i simply want to get a play through working.
Cant post images because new to stackoverflow so heres the link to the image: http://i.stack.imgur.com/RHjlv.jpg
Listing:
This is where I setup the audioReadSettigns which will be used for the AVAssetReaderAudioMixOutput
// Set the read settings
audioReadSettings = [[NSMutableDictionary alloc] init];
[audioReadSettings setValue:[NSNumber numberWithInt:kAudioFormatLinearPCM]
forKey:AVFormatIDKey];
[audioReadSettings setValue:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey];
[audioReadSettings setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey];
[audioReadSettings setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey];
[audioReadSettings setValue:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsNonInterleaved];
[audioReadSettings setValue:[NSNumber numberWithFloat:44100.0] forKey:AVSampleRateKey];
Now the following code listing is a method that receives an NSString with the persistant_id of the song:
-(BOOL)setNextSongID:(NSString*)persistand_id {
assert(persistand_id != nil);
MPMediaItem *song = [self getMediaItemForPersistantID:persistand_id];
NSURL *assetUrl = [song valueForProperty:MPMediaItemPropertyAssetURL];
AVURLAsset *songAsset = [AVURLAsset URLAssetWithURL:assetUrl
options:[NSDictionary dictionaryWithObject:[NSNumber numberWithBool:YES]
forKey:AVURLAssetPreferPreciseDurationAndTimingKey]];
NSError *assetError = nil;
assetReader = [[AVAssetReader assetReaderWithAsset:songAsset error:&assetError] retain];
if (assetError) {
NSLog(#"error: %#", assetError);
return NO;
}
CMTimeRange timeRange = CMTimeRangeMake(kCMTimeZero, songAsset.duration);
[assetReader setTimeRange:timeRange];
track = [[songAsset tracksWithMediaType:AVMediaTypeAudio] objectAtIndex:0];
assetReaderOutput = [AVAssetReaderAudioMixOutput assetReaderAudioMixOutputWithAudioTracks:[NSArray arrayWithObject:track]
audioSettings:audioReadSettings];
if (![assetReader canAddOutput:assetReaderOutput]) {
NSLog(#"cant add reader output... die!");
return NO;
}
[assetReader addOutput:assetReaderOutput];
[assetReader startReading];
// just getting some basic information about the track to print
NSArray *formatDesc = ((AVAssetTrack*)[[assetReaderOutput audioTracks] objectAtIndex:0]).formatDescriptions;
for (unsigned int i = 0; i < [formatDesc count]; ++i) {
CMAudioFormatDescriptionRef item = (CMAudioFormatDescriptionRef)[formatDesc objectAtIndex:i];
const CAStreamBasicDescription *asDesc = (CAStreamBasicDescription*)CMAudioFormatDescriptionGetStreamBasicDescription(item);
if (asDesc) {
// get data
numChannels = asDesc->mChannelsPerFrame;
sampleRate = asDesc->mSampleRate;
asDesc->Print();
}
}
[self copyEnoughSamplesToBufferForLength:24000];
return YES;
}
The following presents the function -(void)copyEnoughSamplesToBufferForLength:
-(void)copyEnoughSamplesToBufferForLength:(UInt32)samples_count {
[w_lock lock];
int stillToCopy = 0;
if (sampleBuffer->numSamples() < samples_count) {
stillToCopy = samples_count;
}
NSAutoreleasePool *apool = [[NSAutoreleasePool alloc] init];
CMSampleBufferRef sampleBufferRef;
SInt16 *dataBuffer = (SInt16*)malloc(8192 * sizeof(SInt16));
int a = 0;
while (stillToCopy > 0) {
sampleBufferRef = [assetReaderOutput copyNextSampleBuffer];
if (!sampleBufferRef) {
// end of song or no more samples
return;
}
CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBufferRef);
CMItemCount numSamplesInBuffer = CMSampleBufferGetNumSamples(sampleBufferRef);
AudioBufferList audioBufferList;
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBufferRef,
NULL,
&audioBufferList,
sizeof(audioBufferList),
NULL,
NULL,
0,
&blockBuffer);
int data_length = floorf(numSamplesInBuffer * 1.0f);
int j = 0;
for (int bufferCount=0; bufferCount < audioBufferList.mNumberBuffers; bufferCount++) {
SInt16* samples = (SInt16 *)audioBufferList.mBuffers[bufferCount].mData;
for (int i=0; i < numSamplesInBuffer; i++) {
dataBuffer[j] = samples[i];
j++;
}
}
CFRelease(sampleBufferRef);
sampleBuffer->putSamples(dataBuffer, j);
stillToCopy = stillToCopy - data_length;
}
free(dataBuffer);
[w_lock unlock];
[apool release];
}
Now the sampleBuffer will have incorrectly decoded samples. Can anyone help me why this is so? This happens for different files on my iTunes library (mp3, aac, wav, etc).
Any help would be greatly appreciated, furthermore, if you need any other listing of my code, or perhaps what the output sounds like, I will attach it per request. I have been sitting on this for the past week trying to debug it and have found no help online -- everyone seems to be doign it in my way, yet it seems that only I have this issue.
Thanks for any help at all!
Peter
Currently, I am also working on a project which involves extracting audio samples from iTunes Library into AudioUnit.
The audiounit render call back is included for your reference. The input format is set as SInt16StereoStreamFormat.
I have made use of Michael Tyson's circular buffer implementation - TPCircularBuffer as the buffer storage. Very easy to use and understand!!! Thanks Michael!
- (void) loadBuffer:(NSURL *)assetURL_
{
if (nil != self.iPodAssetReader) {
[iTunesOperationQueue cancelAllOperations];
[self cleanUpBuffer];
}
NSDictionary *outputSettings = [NSDictionary dictionaryWithObjectsAndKeys:
[NSNumber numberWithInt:kAudioFormatLinearPCM], AVFormatIDKey,
[NSNumber numberWithFloat:44100.0], AVSampleRateKey,
[NSNumber numberWithInt:16], AVLinearPCMBitDepthKey,
[NSNumber numberWithBool:NO], AVLinearPCMIsNonInterleaved,
[NSNumber numberWithBool:NO], AVLinearPCMIsFloatKey,
[NSNumber numberWithBool:NO], AVLinearPCMIsBigEndianKey,
nil];
AVURLAsset *asset = [AVURLAsset URLAssetWithURL:assetURL_ options:nil];
if (asset == nil) {
NSLog(#"asset is not defined!");
return;
}
NSLog(#"Total Asset Duration: %f", CMTimeGetSeconds(asset.duration));
NSError *assetError = nil;
self.iPodAssetReader = [AVAssetReader assetReaderWithAsset:asset error:&assetError];
if (assetError) {
NSLog (#"error: %#", assetError);
return;
}
AVAssetReaderOutput *readerOutput = [AVAssetReaderAudioMixOutput assetReaderAudioMixOutputWithAudioTracks:asset.tracks audioSettings:outputSettings];
if (! [iPodAssetReader canAddOutput: readerOutput]) {
NSLog (#"can't add reader output... die!");
return;
}
// add output reader to reader
[iPodAssetReader addOutput: readerOutput];
if (! [iPodAssetReader startReading]) {
NSLog(#"Unable to start reading!");
return;
}
// Init circular buffer
TPCircularBufferInit(&playbackState.circularBuffer, kTotalBufferSize);
__block NSBlockOperation * feediPodBufferOperation = [NSBlockOperation blockOperationWithBlock:^{
while (![feediPodBufferOperation isCancelled] && iPodAssetReader.status != AVAssetReaderStatusCompleted) {
if (iPodAssetReader.status == AVAssetReaderStatusReading) {
// Check if the available buffer space is enough to hold at least one cycle of the sample data
if (kTotalBufferSize - playbackState.circularBuffer.fillCount >= 32768) {
CMSampleBufferRef nextBuffer = [readerOutput copyNextSampleBuffer];
if (nextBuffer) {
AudioBufferList abl;
CMBlockBufferRef blockBuffer;
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(nextBuffer, NULL, &abl, sizeof(abl), NULL, NULL, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &blockBuffer);
UInt64 size = CMSampleBufferGetTotalSampleSize(nextBuffer);
int bytesCopied = TPCircularBufferProduceBytes(&playbackState.circularBuffer, abl.mBuffers[0].mData, size);
if (!playbackState.bufferIsReady && bytesCopied > 0) {
playbackState.bufferIsReady = YES;
}
CFRelease(nextBuffer);
CFRelease(blockBuffer);
}
else {
break;
}
}
}
}
NSLog(#"iPod Buffer Reading Finished");
}];
[iTunesOperationQueue addOperation:feediPodBufferOperation];
}
static OSStatus ipodRenderCallback (
void *inRefCon, // A pointer to a struct containing the complete audio data
// to play, as well as state information such as the
// first sample to play on this invocation of the callback.
AudioUnitRenderActionFlags *ioActionFlags, // Unused here. When generating audio, use ioActionFlags to indicate silence
// between sounds; for silence, also memset the ioData buffers to 0.
const AudioTimeStamp *inTimeStamp, // Unused here.
UInt32 inBusNumber, // The mixer unit input bus that is requesting some new
// frames of audio data to play.
UInt32 inNumberFrames, // The number of frames of audio to provide to the buffer(s)
// pointed to by the ioData parameter.
AudioBufferList *ioData // On output, the audio data to play. The callback's primary
// responsibility is to fill the buffer(s) in the
// AudioBufferList.
)
{
Audio* audioObject = (Audio*)inRefCon;
AudioSampleType *outSample = (AudioSampleType *)ioData->mBuffers[0].mData;
// Zero-out all the output samples first
memset(outSample, 0, inNumberFrames * kUnitSize * 2);
if ( audioObject.playingiPod && audioObject.bufferIsReady) {
// Pull audio from circular buffer
int32_t availableBytes;
AudioSampleType *bufferTail = TPCircularBufferTail(&audioObject.circularBuffer, &availableBytes);
memcpy(outSample, bufferTail, MIN(availableBytes, inNumberFrames * kUnitSize * 2) );
TPCircularBufferConsume(&audioObject.circularBuffer, MIN(availableBytes, inNumberFrames * kUnitSize * 2) );
audioObject.currentSampleNum += MIN(availableBytes / (kUnitSize * 2), inNumberFrames);
if (availableBytes <= inNumberFrames * kUnitSize * 2) {
// Buffer is running out or playback is finished
audioObject.bufferIsReady = NO;
audioObject.playingiPod = NO;
audioObject.currentSampleNum = 0;
if ([[audioObject delegate] respondsToSelector:#selector(playbackDidFinish)]) {
[[audioObject delegate] performSelector:#selector(playbackDidFinish)];
}
}
}
return noErr;
}
- (void) setupSInt16StereoStreamFormat {
// The AudioUnitSampleType data type is the recommended type for sample data in audio
// units. This obtains the byte size of the type for use in filling in the ASBD.
size_t bytesPerSample = sizeof (AudioSampleType);
// Fill the application audio format struct's fields to define a linear PCM,
// stereo, noninterleaved stream at the hardware sample rate.
SInt16StereoStreamFormat.mFormatID = kAudioFormatLinearPCM;
SInt16StereoStreamFormat.mFormatFlags = kAudioFormatFlagsCanonical;
SInt16StereoStreamFormat.mBytesPerPacket = 2 * bytesPerSample; // *** kAudioFormatFlagsCanonical <- implicit interleaved data => (left sample + right sample) per Packet
SInt16StereoStreamFormat.mFramesPerPacket = 1;
SInt16StereoStreamFormat.mBytesPerFrame = SInt16StereoStreamFormat.mBytesPerPacket * SInt16StereoStreamFormat.mFramesPerPacket;
SInt16StereoStreamFormat.mChannelsPerFrame = 2; // 2 indicates stereo
SInt16StereoStreamFormat.mBitsPerChannel = 8 * bytesPerSample;
SInt16StereoStreamFormat.mSampleRate = graphSampleRate;
NSLog (#"The stereo stream format for the \"iPod\" mixer input bus:");
[self printASBD: SInt16StereoStreamFormat];
}
I guess it is kind of late, but you could try this library:
https://bitbucket.org/artgillespie/tslibraryimport
After using this to save the audio into a file, you could process the data with render callbacks from MixerHost.
If I were you I would either use kAudioUnitSubType_AudioFilePlayer to play the file and access its samples with the units render callback.
Or
Use ExtAudioFileRef to extract the samples straight to a buffer.