Related
From a delegate method I am receiving an AudioBufferList while I am recording audio. I am trying to collect the data from the AudioBufferList and save it so I can load it into my AVAudioPlayer but the AVAudioPlayer throws an error and I am unable to play the recording. I need to be able to play the audio through AVAudioPlayer without having the file, just by using the AudioBufferList.
Originally I was saving the recording to a file then loading it into AVAudioPlayer but with this method I was unable to append to the recording without having to make another audio file then merging the 2 files after the append was made. This was taking to much time and I would still like to be able to listen to the recording between appends. So now I am not saving the audio file so that I can keep appending to it until I wish to save it. The problem with this is the NSData that I am saving from the AudioBufferList is not loading into the AVAudioPlayer properly.
Here is my code for gathering the NSData:
- (void) microphone:(EZMicrophone *)microphone
hasBufferList:(AudioBufferList *)bufferList
withBufferSize:(UInt32)bufferSize
withNumberOfChannels:(UInt32)numberOfChannels
{
AudioBuffer sourceBuffer = bufferList->mBuffers[0];
if (audioBuffer.mDataByteSize != sourceBuffer.mDataByteSize)
{
free(audioBuffer.mData);
audioBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
audioBuffer.mData = malloc(sourceBuffer.mDataByteSize);
}
int currentBuffer =0;
int maxBuf = 800;
for( int y=0; y< bufferList->mNumberBuffers; y++ )
{
if (currentBuffer < maxBuf)
{
AudioBuffer audioBuff = bufferList->mBuffers[y];
Float32 *frame = (Float32*)audioBuff.mData;
[data appendBytes:frame length:audioBuffer.mDataByteSize];
currentBuffer += audioBuff.mDataByteSize;
}
else
{
break;
}
}
}
When I try and load the NSData into AVAudioPlayer I get the following error:
self.audioPlayer = [[AVAudioPlayer alloc] initWithData:data error:&err];
err:
Error Domain=NSOSStatusErrorDomain Code=1954115647 "The operation couldn’t be completed. (OSStatus error 1954115647.)"
Any help would be appreciated.
Thank you,
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?
Is it theoretically possible to record a phone call on iPhone?
I'm accepting answers which:
may or may not require the phone to be jailbroken
may or may not pass apple's guidelines due to use of private API's (I don't care; it is not for the App Store)
may or may not use private SDKs
I don't want answers just bluntly saying "Apple does not allow that".
I know there would be no official way of doing it, and certainly not for an App Store application, and I know there are call recording apps which place outgoing calls through their own servers.
Here you go. Complete working example. Tweak should be loaded in mediaserverd daemon. It will record every phone call in /var/mobile/Media/DCIM/result.m4a. Audio file has two channels. Left is microphone, right is speaker. On iPhone 4S call is recorded only when the speaker is turned on. On iPhone 5, 5C and 5S call is recorded either way. There might be small hiccups when switching to/from speaker but recording will continue.
#import <AudioToolbox/AudioToolbox.h>
#import <libkern/OSAtomic.h>
//CoreTelephony.framework
extern "C" CFStringRef const kCTCallStatusChangeNotification;
extern "C" CFStringRef const kCTCallStatus;
extern "C" id CTTelephonyCenterGetDefault();
extern "C" void CTTelephonyCenterAddObserver(id ct, void* observer, CFNotificationCallback callBack, CFStringRef name, void *object, CFNotificationSuspensionBehavior sb);
extern "C" int CTGetCurrentCallCount();
enum
{
kCTCallStatusActive = 1,
kCTCallStatusHeld = 2,
kCTCallStatusOutgoing = 3,
kCTCallStatusIncoming = 4,
kCTCallStatusHanged = 5
};
NSString* kMicFilePath = #"/var/mobile/Media/DCIM/mic.caf";
NSString* kSpeakerFilePath = #"/var/mobile/Media/DCIM/speaker.caf";
NSString* kResultFilePath = #"/var/mobile/Media/DCIM/result.m4a";
OSSpinLock phoneCallIsActiveLock = 0;
OSSpinLock speakerLock = 0;
OSSpinLock micLock = 0;
ExtAudioFileRef micFile = NULL;
ExtAudioFileRef speakerFile = NULL;
BOOL phoneCallIsActive = NO;
void Convert()
{
//File URLs
CFURLRef micUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kMicFilePath, kCFURLPOSIXPathStyle, false);
CFURLRef speakerUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kSpeakerFilePath, kCFURLPOSIXPathStyle, false);
CFURLRef mixUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kResultFilePath, kCFURLPOSIXPathStyle, false);
ExtAudioFileRef micFile = NULL;
ExtAudioFileRef speakerFile = NULL;
ExtAudioFileRef mixFile = NULL;
//Opening input files (speaker and mic)
ExtAudioFileOpenURL(micUrl, &micFile);
ExtAudioFileOpenURL(speakerUrl, &speakerFile);
//Reading input file audio format (mono LPCM)
AudioStreamBasicDescription inputFormat, outputFormat;
UInt32 descSize = sizeof(inputFormat);
ExtAudioFileGetProperty(micFile, kExtAudioFileProperty_FileDataFormat, &descSize, &inputFormat);
int sampleSize = inputFormat.mBytesPerFrame;
//Filling input stream format for output file (stereo LPCM)
FillOutASBDForLPCM(inputFormat, inputFormat.mSampleRate, 2, inputFormat.mBitsPerChannel, inputFormat.mBitsPerChannel, true, false, false);
//Filling output file audio format (AAC)
memset(&outputFormat, 0, sizeof(outputFormat));
outputFormat.mFormatID = kAudioFormatMPEG4AAC;
outputFormat.mSampleRate = 8000;
outputFormat.mFormatFlags = kMPEG4Object_AAC_Main;
outputFormat.mChannelsPerFrame = 2;
//Opening output file
ExtAudioFileCreateWithURL(mixUrl, kAudioFileM4AType, &outputFormat, NULL, kAudioFileFlags_EraseFile, &mixFile);
ExtAudioFileSetProperty(mixFile, kExtAudioFileProperty_ClientDataFormat, sizeof(inputFormat), &inputFormat);
//Freeing URLs
CFRelease(micUrl);
CFRelease(speakerUrl);
CFRelease(mixUrl);
//Setting up audio buffers
int bufferSizeInSamples = 64 * 1024;
AudioBufferList micBuffer;
micBuffer.mNumberBuffers = 1;
micBuffer.mBuffers[0].mNumberChannels = 1;
micBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples;
micBuffer.mBuffers[0].mData = malloc(micBuffer.mBuffers[0].mDataByteSize);
AudioBufferList speakerBuffer;
speakerBuffer.mNumberBuffers = 1;
speakerBuffer.mBuffers[0].mNumberChannels = 1;
speakerBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples;
speakerBuffer.mBuffers[0].mData = malloc(speakerBuffer.mBuffers[0].mDataByteSize);
AudioBufferList mixBuffer;
mixBuffer.mNumberBuffers = 1;
mixBuffer.mBuffers[0].mNumberChannels = 2;
mixBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples * 2;
mixBuffer.mBuffers[0].mData = malloc(mixBuffer.mBuffers[0].mDataByteSize);
//Converting
while (true)
{
//Reading data from input files
UInt32 framesToRead = bufferSizeInSamples;
ExtAudioFileRead(micFile, &framesToRead, &micBuffer);
ExtAudioFileRead(speakerFile, &framesToRead, &speakerBuffer);
if (framesToRead == 0)
{
break;
}
//Building interleaved stereo buffer - left channel is mic, right - speaker
for (int i = 0; i < framesToRead; i++)
{
memcpy((char*)mixBuffer.mBuffers[0].mData + i * sampleSize * 2, (char*)micBuffer.mBuffers[0].mData + i * sampleSize, sampleSize);
memcpy((char*)mixBuffer.mBuffers[0].mData + i * sampleSize * 2 + sampleSize, (char*)speakerBuffer.mBuffers[0].mData + i * sampleSize, sampleSize);
}
//Writing to output file - LPCM will be converted to AAC
ExtAudioFileWrite(mixFile, framesToRead, &mixBuffer);
}
//Closing files
ExtAudioFileDispose(micFile);
ExtAudioFileDispose(speakerFile);
ExtAudioFileDispose(mixFile);
//Freeing audio buffers
free(micBuffer.mBuffers[0].mData);
free(speakerBuffer.mBuffers[0].mData);
free(mixBuffer.mBuffers[0].mData);
}
void Cleanup()
{
[[NSFileManager defaultManager] removeItemAtPath:kMicFilePath error:NULL];
[[NSFileManager defaultManager] removeItemAtPath:kSpeakerFilePath error:NULL];
}
void CoreTelephonyNotificationCallback(CFNotificationCenterRef center, void *observer, CFStringRef name, const void *object, CFDictionaryRef userInfo)
{
NSDictionary* data = (NSDictionary*)userInfo;
if ([(NSString*)name isEqualToString:(NSString*)kCTCallStatusChangeNotification])
{
int currentCallStatus = [data[(NSString*)kCTCallStatus] integerValue];
if (currentCallStatus == kCTCallStatusActive)
{
OSSpinLockLock(&phoneCallIsActiveLock);
phoneCallIsActive = YES;
OSSpinLockUnlock(&phoneCallIsActiveLock);
}
else if (currentCallStatus == kCTCallStatusHanged)
{
if (CTGetCurrentCallCount() > 0)
{
return;
}
OSSpinLockLock(&phoneCallIsActiveLock);
phoneCallIsActive = NO;
OSSpinLockUnlock(&phoneCallIsActiveLock);
//Closing mic file
OSSpinLockLock(&micLock);
if (micFile != NULL)
{
ExtAudioFileDispose(micFile);
}
micFile = NULL;
OSSpinLockUnlock(&micLock);
//Closing speaker file
OSSpinLockLock(&speakerLock);
if (speakerFile != NULL)
{
ExtAudioFileDispose(speakerFile);
}
speakerFile = NULL;
OSSpinLockUnlock(&speakerLock);
Convert();
Cleanup();
}
}
}
OSStatus(*AudioUnitProcess_orig)(AudioUnit unit, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inNumberFrames, AudioBufferList *ioData);
OSStatus AudioUnitProcess_hook(AudioUnit unit, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inNumberFrames, AudioBufferList *ioData)
{
OSSpinLockLock(&phoneCallIsActiveLock);
if (phoneCallIsActive == NO)
{
OSSpinLockUnlock(&phoneCallIsActiveLock);
return AudioUnitProcess_orig(unit, ioActionFlags, inTimeStamp, inNumberFrames, ioData);
}
OSSpinLockUnlock(&phoneCallIsActiveLock);
ExtAudioFileRef* currentFile = NULL;
OSSpinLock* currentLock = NULL;
AudioComponentDescription unitDescription = {0};
AudioComponentGetDescription(AudioComponentInstanceGetComponent(unit), &unitDescription);
//'agcc', 'mbdp' - iPhone 4S, iPhone 5
//'agc2', 'vrq2' - iPhone 5C, iPhone 5S
if (unitDescription.componentSubType == 'agcc' || unitDescription.componentSubType == 'agc2')
{
currentFile = &micFile;
currentLock = &micLock;
}
else if (unitDescription.componentSubType == 'mbdp' || unitDescription.componentSubType == 'vrq2')
{
currentFile = &speakerFile;
currentLock = &speakerLock;
}
if (currentFile != NULL)
{
OSSpinLockLock(currentLock);
//Opening file
if (*currentFile == NULL)
{
//Obtaining input audio format
AudioStreamBasicDescription desc;
UInt32 descSize = sizeof(desc);
AudioUnitGetProperty(unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &desc, &descSize);
//Opening audio file
CFURLRef url = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)((currentFile == &micFile) ? kMicFilePath : kSpeakerFilePath), kCFURLPOSIXPathStyle, false);
ExtAudioFileRef audioFile = NULL;
OSStatus result = ExtAudioFileCreateWithURL(url, kAudioFileCAFType, &desc, NULL, kAudioFileFlags_EraseFile, &audioFile);
if (result != 0)
{
*currentFile = NULL;
}
else
{
*currentFile = audioFile;
//Writing audio format
ExtAudioFileSetProperty(*currentFile, kExtAudioFileProperty_ClientDataFormat, sizeof(desc), &desc);
}
CFRelease(url);
}
else
{
//Writing audio buffer
ExtAudioFileWrite(*currentFile, inNumberFrames, ioData);
}
OSSpinLockUnlock(currentLock);
}
return AudioUnitProcess_orig(unit, ioActionFlags, inTimeStamp, inNumberFrames, ioData);
}
__attribute__((constructor))
static void initialize()
{
CTTelephonyCenterAddObserver(CTTelephonyCenterGetDefault(), NULL, CoreTelephonyNotificationCallback, NULL, NULL, CFNotificationSuspensionBehaviorHold);
MSHookFunction(AudioUnitProcess, AudioUnitProcess_hook, &AudioUnitProcess_orig);
}
A few words about what's going on. AudioUnitProcess function is used for processing audio streams in order to apply some effects, mix, convert etc. We are hooking AudioUnitProcess in order to access phone call's audio streams. While phone call is active these streams are being processed in various ways.
We are listening for CoreTelephony notifications in order to get phone call status changes. When we receive audio samples we need to determine where they come from - microphone or speaker. This is done using componentSubType field in AudioComponentDescription structure. Now, you might think, why don't we store AudioUnit objects so that we don't need to check componentSubType every time. I did that but it will break everything when you switch speaker on/off on iPhone 5 because AudioUnit objects will change, they are recreated. So, now we open audio files (one for microphone and one for speaker) and write samples in them, simple as that. When phone call ends we will receive appropriate CoreTelephony notification and close the files. We have two separate files with audio from microphone and speaker that we need to merge. This is what void Convert() is for. It's pretty simple if you know the API. I don't think I need to explain it, comments are enough.
About locks. There are many threads in mediaserverd. Audio processing and CoreTelephony notifications are on different threads so we need some kind synchronization. I chose spin locks because they are fast and because the chance of lock contention is small in our case. On iPhone 4S and even iPhone 5 all the work in AudioUnitProcess should be done as fast as possible otherwise you will hear hiccups from device speaker which obviously not good.
Yes. Audio Recorder by a developer named Limneos does that (and quite well). You can find it on Cydia. It can record any type of call on iPhone 5 and up without using any servers etc'. The call will be placed on the device in an Audio file. It also supports iPhone 4S but for speaker only.
This tweak is known to be the first tweak ever that managed to record both streams of audio without using any 3rd party severs, VOIP or something similar.
The developer placed beeps on the other side of the call to alert the person you are recording but those were removed too by hackers across the net. To answer your question, Yes, it's very much possible, and not just theoretically.
Further reading
https://stackoverflow.com/a/19413363/202451
http://forums.macrumors.com/showthread.php?t=1566350
https://github.com/nst/iOS-Runtime-Headers
The only solution I can think of is to use the Core Telephony framework, and more specifically the callEventHandler property, to intercept when a call is coming in, and then to use an AVAudioRecorder to record the voice of the person with the phone (and maybe a little of the person on the other line's voice). This is obviously not perfect, and would only work if your application is in the foreground at the time of the call, but it may be the best you can get. See more about finding out if there is an incoming phone call here: Can we fire an event when ever there is Incoming and Outgoing call in iphone?.
EDIT:
.h:
#import <AVFoundation/AVFoundation.h>
#import<CoreTelephony/CTCallCenter.h>
#import<CoreTelephony/CTCall.h>
#property (strong, nonatomic) AVAudioRecorder *audioRecorder;
ViewDidLoad:
NSArray *dirPaths;
NSString *docsDir;
dirPaths = NSSearchPathForDirectoriesInDomains(
NSDocumentDirectory, NSUserDomainMask, YES);
docsDir = dirPaths[0];
NSString *soundFilePath = [docsDir
stringByAppendingPathComponent:#"sound.caf"];
NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath];
NSDictionary *recordSettings = [NSDictionary
dictionaryWithObjectsAndKeys:
[NSNumber numberWithInt:AVAudioQualityMin],
AVEncoderAudioQualityKey,
[NSNumber numberWithInt:16],
AVEncoderBitRateKey,
[NSNumber numberWithInt: 2],
AVNumberOfChannelsKey,
[NSNumber numberWithFloat:44100.0],
AVSampleRateKey,
nil];
NSError *error = nil;
_audioRecorder = [[AVAudioRecorder alloc]
initWithURL:soundFileURL
settings:recordSettings
error:&error];
if (error)
{
NSLog(#"error: %#", [error localizedDescription]);
} else {
[_audioRecorder prepareToRecord];
}
CTCallCenter *callCenter = [[CTCallCenter alloc] init];
[callCenter setCallEventHandler:^(CTCall *call) {
if ([[call callState] isEqual:CTCallStateConnected]) {
[_audioRecorder record];
} else if ([[call callState] isEqual:CTCallStateDisconnected]) {
[_audioRecorder stop];
}
}];
AppDelegate.m:
- (void)applicationDidEnterBackground:(UIApplication *)application//Makes sure that the recording keeps happening even when app is in the background, though only can go for 10 minutes.
{
__block UIBackgroundTaskIdentifier task = 0;
task=[application beginBackgroundTaskWithExpirationHandler:^{
NSLog(#"Expiration handler called %f",[application backgroundTimeRemaining]);
[application endBackgroundTask:task];
task=UIBackgroundTaskInvalid;
}];
This is the first time using many of these features, so not sure if this is exactly right, but I think you get the idea. Untested, as I do not have access to the right tools at the moment. Compiled using these sources:
Recording voice in background using AVAudioRecorder
http://prassan-warrior.blogspot.com/2012/11/recording-audio-on-iphone-with.html
Can we fire an event when ever there is Incoming and Outgoing call in iphone?
Apple does not allow it and does not provide any API for it.
However, on a jailbroken device I'm sure it's possible. As a matter of fact, I think it's already done. I remember seeing an app when my phone was jailbroken that changed your voice and recorded the call - I remember it was a US company offering it only in the states. Unfortunately I don't remember the name...
I guess some hardware could solve this. Connected to the minijack-port; having earbuds and a microphone passing through a small recorder. This recorder can be very simple. While not in conversation the recorder could feed the phone with data/the recording (through the jack-cable). And with a simple start button (just like the volum controls on the earbuds) could be sufficient for timing the recording.
Some setups
http://www.danmccomb.com/posts/483/how-to-record-iphone-conversations-using-zoom-h4n/
http://forums.macrumors.com/showthread.php?t=346430
based on my research.. people keep on saying that it's based on mismatched/wrong formatting.. but i'm using lPCM formatting for both input and output.. how can you go wrong with that? the result i'm getting is just noise.. (like white noise)
I've decided to just paste my entire code.. perhaps that would help:
#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
Update:
btomw's answer below solved the problem magnificently. But I want to get to the bottom of this (most novice developers like myself and even btomw when he first started usually shoot in the dark with parameters, formatting etc - see here for an example -)..
the reason why I provided nul as a parameter for
AVURLAsset *songAsset = [AVURLAsset URLAssetWithURL:assetURL options:audioReadSettings];
was because according to the documentation and trial and error, I realized that any formatting I put other than lPCM would be rejected outright. In other words, when you use AVAseetReader or conversion even the result is always lPCM.. so I thought the default format was lPCM anyways and so I left it as null.. but I guess I was wrong.
The weird part in this (please correct me anyone, if I'm wrong) is that as I mentioned.. supposed the original file was .mp3, and my intention was to play it back (or send the packets over a network etc) as mp3.. and so I provided an mp3 ABSD.. the asset reader will crash! so is that if i wanted to send it in it's original form, i just supply null? the obvious problem with this is that there would be no way for me to figure out what ABSD it has once I receive it on the other side.. or could I?
Update 2:You can download the code from github.
So here's what I think is happening and also how I think you can fix it.
You're pulling a predefined item out of the ipod (music) library on an iOS device. you are then using an asset reader to collect it's buffers, and queue those buffers, where possible, in an AudioQueue.
The problem you are having, I think, is that you are setting the audio queue buffer's input format to Linear Pulse Code Modulation (LPCM - hope I got that right, I might be off on the acronym). The output settings you are passing to the asset reader output are nil, which means that you'll get an output that is most likely NOT LPCM, but is instead aiff or aac or mp3 or whatever the format is of the song as it exists in iOS's media library. You can, however, remedy this situation by passing in different output settings.
Try changing
readerOutput = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:track outputSettings:nil];
to:
[NSDictionary dictionaryWithObjectsAndKeys:
[NSNumber numberWithInt:kAudioFormatLinearPCM], AVFormatIDKey,
[NSNumber numberWithFloat:44100.0], AVSampleRateKey,
[NSNumber numberWithInt:2], AVNumberOfChannelsKey,
[NSData dataWithBytes:&channelLayout length:sizeof(AudioChannelLayout)],
AVChannelLayoutKey,
[NSNumber numberWithInt:16], AVLinearPCMBitDepthKey,
[NSNumber numberWithBool:NO], AVLinearPCMIsNonInterleaved,
[NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
[NSNumber numberWithBool:NO], AVLinearPCMIsBigEndianKey,
nil];
output = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:track audioSettings:outputSettings];
It's my understanding (per the documentation at Apple1) that passing nil as the output settings param gives you samples of the same file type as the original audio track. Even if you have a file that is LPCM, some other settings might be off, which might cause your problems. At the very least, this will normalize all the reader output, which should make things a bit easier to trouble shoot.
Hope that helps!
Edit:
the reason why I provided nul as a parameter for AVURLAsset *songAsset
= [AVURLAsset URLAssetWithURL:assetURL options:audioReadSettings];
was because according to the documentation and trial and error, I...
AVAssetReaders do 2 things; read back an audio file as it exists on disk (i.e.: mp3, aac, aiff), or convert the audio into lpcm.
If you pass nil as the output settings, it will read the file back as it exists, and in this you are correct. I apologize for not mentioning that an asset reader will only allow nil or LPCM. I actually ran into that problem myself (it's in the docs somewhere, but requires a bit of diving), but didn't elect to mention it here as it wasn't on my mind at the time. Sooooo... sorry about that?
If you want to know the AudioStreamBasicDescription (ASBD) of the track you are reading before you read it, you can get it by doing this:
AVURLAsset* uasset = [[AVURLAsset URLAssetWithURL:<#assetURL#> options:nil]retain];
AVAssetTrack*track = [uasset.tracks objectAtIndex:0];
CMFormatDescriptionRef formDesc = (CMFormatDescriptionRef)[[track formatDescriptions] objectAtIndex:0];
const AudioStreamBasicDescription* asbdPointer = CMAudioFormatDescriptionGetStreamBasicDescription(formDesc);
//because this is a pointer and not a struct we need to move the data into a struct so we can use it
AudioStreamBasicDescription asbd = {0};
memcpy(&asbd, asbdPointer, sizeof(asbd));
//asbd now contains a basic description for the track
You can then convert asbd to binary data in whatever format you see fit and transfer it over the network. You should then be able to start sending audio buffer data over the network and successfully play it back with your AudioQueue.
I actually had a system like this working not that long ago, but since I could't keep the connection alive when the iOS client device went to the background, I wasn't able to use it for my purpose. Still, if all that work lets me help someone else who can actually use the info, seems like a win to me.
Is it theoretically possible to record a phone call on iPhone?
I'm accepting answers which:
may or may not require the phone to be jailbroken
may or may not pass apple's guidelines due to use of private API's (I don't care; it is not for the App Store)
may or may not use private SDKs
I don't want answers just bluntly saying "Apple does not allow that".
I know there would be no official way of doing it, and certainly not for an App Store application, and I know there are call recording apps which place outgoing calls through their own servers.
Here you go. Complete working example. Tweak should be loaded in mediaserverd daemon. It will record every phone call in /var/mobile/Media/DCIM/result.m4a. Audio file has two channels. Left is microphone, right is speaker. On iPhone 4S call is recorded only when the speaker is turned on. On iPhone 5, 5C and 5S call is recorded either way. There might be small hiccups when switching to/from speaker but recording will continue.
#import <AudioToolbox/AudioToolbox.h>
#import <libkern/OSAtomic.h>
//CoreTelephony.framework
extern "C" CFStringRef const kCTCallStatusChangeNotification;
extern "C" CFStringRef const kCTCallStatus;
extern "C" id CTTelephonyCenterGetDefault();
extern "C" void CTTelephonyCenterAddObserver(id ct, void* observer, CFNotificationCallback callBack, CFStringRef name, void *object, CFNotificationSuspensionBehavior sb);
extern "C" int CTGetCurrentCallCount();
enum
{
kCTCallStatusActive = 1,
kCTCallStatusHeld = 2,
kCTCallStatusOutgoing = 3,
kCTCallStatusIncoming = 4,
kCTCallStatusHanged = 5
};
NSString* kMicFilePath = #"/var/mobile/Media/DCIM/mic.caf";
NSString* kSpeakerFilePath = #"/var/mobile/Media/DCIM/speaker.caf";
NSString* kResultFilePath = #"/var/mobile/Media/DCIM/result.m4a";
OSSpinLock phoneCallIsActiveLock = 0;
OSSpinLock speakerLock = 0;
OSSpinLock micLock = 0;
ExtAudioFileRef micFile = NULL;
ExtAudioFileRef speakerFile = NULL;
BOOL phoneCallIsActive = NO;
void Convert()
{
//File URLs
CFURLRef micUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kMicFilePath, kCFURLPOSIXPathStyle, false);
CFURLRef speakerUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kSpeakerFilePath, kCFURLPOSIXPathStyle, false);
CFURLRef mixUrl = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)kResultFilePath, kCFURLPOSIXPathStyle, false);
ExtAudioFileRef micFile = NULL;
ExtAudioFileRef speakerFile = NULL;
ExtAudioFileRef mixFile = NULL;
//Opening input files (speaker and mic)
ExtAudioFileOpenURL(micUrl, &micFile);
ExtAudioFileOpenURL(speakerUrl, &speakerFile);
//Reading input file audio format (mono LPCM)
AudioStreamBasicDescription inputFormat, outputFormat;
UInt32 descSize = sizeof(inputFormat);
ExtAudioFileGetProperty(micFile, kExtAudioFileProperty_FileDataFormat, &descSize, &inputFormat);
int sampleSize = inputFormat.mBytesPerFrame;
//Filling input stream format for output file (stereo LPCM)
FillOutASBDForLPCM(inputFormat, inputFormat.mSampleRate, 2, inputFormat.mBitsPerChannel, inputFormat.mBitsPerChannel, true, false, false);
//Filling output file audio format (AAC)
memset(&outputFormat, 0, sizeof(outputFormat));
outputFormat.mFormatID = kAudioFormatMPEG4AAC;
outputFormat.mSampleRate = 8000;
outputFormat.mFormatFlags = kMPEG4Object_AAC_Main;
outputFormat.mChannelsPerFrame = 2;
//Opening output file
ExtAudioFileCreateWithURL(mixUrl, kAudioFileM4AType, &outputFormat, NULL, kAudioFileFlags_EraseFile, &mixFile);
ExtAudioFileSetProperty(mixFile, kExtAudioFileProperty_ClientDataFormat, sizeof(inputFormat), &inputFormat);
//Freeing URLs
CFRelease(micUrl);
CFRelease(speakerUrl);
CFRelease(mixUrl);
//Setting up audio buffers
int bufferSizeInSamples = 64 * 1024;
AudioBufferList micBuffer;
micBuffer.mNumberBuffers = 1;
micBuffer.mBuffers[0].mNumberChannels = 1;
micBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples;
micBuffer.mBuffers[0].mData = malloc(micBuffer.mBuffers[0].mDataByteSize);
AudioBufferList speakerBuffer;
speakerBuffer.mNumberBuffers = 1;
speakerBuffer.mBuffers[0].mNumberChannels = 1;
speakerBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples;
speakerBuffer.mBuffers[0].mData = malloc(speakerBuffer.mBuffers[0].mDataByteSize);
AudioBufferList mixBuffer;
mixBuffer.mNumberBuffers = 1;
mixBuffer.mBuffers[0].mNumberChannels = 2;
mixBuffer.mBuffers[0].mDataByteSize = sampleSize * bufferSizeInSamples * 2;
mixBuffer.mBuffers[0].mData = malloc(mixBuffer.mBuffers[0].mDataByteSize);
//Converting
while (true)
{
//Reading data from input files
UInt32 framesToRead = bufferSizeInSamples;
ExtAudioFileRead(micFile, &framesToRead, &micBuffer);
ExtAudioFileRead(speakerFile, &framesToRead, &speakerBuffer);
if (framesToRead == 0)
{
break;
}
//Building interleaved stereo buffer - left channel is mic, right - speaker
for (int i = 0; i < framesToRead; i++)
{
memcpy((char*)mixBuffer.mBuffers[0].mData + i * sampleSize * 2, (char*)micBuffer.mBuffers[0].mData + i * sampleSize, sampleSize);
memcpy((char*)mixBuffer.mBuffers[0].mData + i * sampleSize * 2 + sampleSize, (char*)speakerBuffer.mBuffers[0].mData + i * sampleSize, sampleSize);
}
//Writing to output file - LPCM will be converted to AAC
ExtAudioFileWrite(mixFile, framesToRead, &mixBuffer);
}
//Closing files
ExtAudioFileDispose(micFile);
ExtAudioFileDispose(speakerFile);
ExtAudioFileDispose(mixFile);
//Freeing audio buffers
free(micBuffer.mBuffers[0].mData);
free(speakerBuffer.mBuffers[0].mData);
free(mixBuffer.mBuffers[0].mData);
}
void Cleanup()
{
[[NSFileManager defaultManager] removeItemAtPath:kMicFilePath error:NULL];
[[NSFileManager defaultManager] removeItemAtPath:kSpeakerFilePath error:NULL];
}
void CoreTelephonyNotificationCallback(CFNotificationCenterRef center, void *observer, CFStringRef name, const void *object, CFDictionaryRef userInfo)
{
NSDictionary* data = (NSDictionary*)userInfo;
if ([(NSString*)name isEqualToString:(NSString*)kCTCallStatusChangeNotification])
{
int currentCallStatus = [data[(NSString*)kCTCallStatus] integerValue];
if (currentCallStatus == kCTCallStatusActive)
{
OSSpinLockLock(&phoneCallIsActiveLock);
phoneCallIsActive = YES;
OSSpinLockUnlock(&phoneCallIsActiveLock);
}
else if (currentCallStatus == kCTCallStatusHanged)
{
if (CTGetCurrentCallCount() > 0)
{
return;
}
OSSpinLockLock(&phoneCallIsActiveLock);
phoneCallIsActive = NO;
OSSpinLockUnlock(&phoneCallIsActiveLock);
//Closing mic file
OSSpinLockLock(&micLock);
if (micFile != NULL)
{
ExtAudioFileDispose(micFile);
}
micFile = NULL;
OSSpinLockUnlock(&micLock);
//Closing speaker file
OSSpinLockLock(&speakerLock);
if (speakerFile != NULL)
{
ExtAudioFileDispose(speakerFile);
}
speakerFile = NULL;
OSSpinLockUnlock(&speakerLock);
Convert();
Cleanup();
}
}
}
OSStatus(*AudioUnitProcess_orig)(AudioUnit unit, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inNumberFrames, AudioBufferList *ioData);
OSStatus AudioUnitProcess_hook(AudioUnit unit, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inNumberFrames, AudioBufferList *ioData)
{
OSSpinLockLock(&phoneCallIsActiveLock);
if (phoneCallIsActive == NO)
{
OSSpinLockUnlock(&phoneCallIsActiveLock);
return AudioUnitProcess_orig(unit, ioActionFlags, inTimeStamp, inNumberFrames, ioData);
}
OSSpinLockUnlock(&phoneCallIsActiveLock);
ExtAudioFileRef* currentFile = NULL;
OSSpinLock* currentLock = NULL;
AudioComponentDescription unitDescription = {0};
AudioComponentGetDescription(AudioComponentInstanceGetComponent(unit), &unitDescription);
//'agcc', 'mbdp' - iPhone 4S, iPhone 5
//'agc2', 'vrq2' - iPhone 5C, iPhone 5S
if (unitDescription.componentSubType == 'agcc' || unitDescription.componentSubType == 'agc2')
{
currentFile = &micFile;
currentLock = &micLock;
}
else if (unitDescription.componentSubType == 'mbdp' || unitDescription.componentSubType == 'vrq2')
{
currentFile = &speakerFile;
currentLock = &speakerLock;
}
if (currentFile != NULL)
{
OSSpinLockLock(currentLock);
//Opening file
if (*currentFile == NULL)
{
//Obtaining input audio format
AudioStreamBasicDescription desc;
UInt32 descSize = sizeof(desc);
AudioUnitGetProperty(unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &desc, &descSize);
//Opening audio file
CFURLRef url = CFURLCreateWithFileSystemPath(NULL, (CFStringRef)((currentFile == &micFile) ? kMicFilePath : kSpeakerFilePath), kCFURLPOSIXPathStyle, false);
ExtAudioFileRef audioFile = NULL;
OSStatus result = ExtAudioFileCreateWithURL(url, kAudioFileCAFType, &desc, NULL, kAudioFileFlags_EraseFile, &audioFile);
if (result != 0)
{
*currentFile = NULL;
}
else
{
*currentFile = audioFile;
//Writing audio format
ExtAudioFileSetProperty(*currentFile, kExtAudioFileProperty_ClientDataFormat, sizeof(desc), &desc);
}
CFRelease(url);
}
else
{
//Writing audio buffer
ExtAudioFileWrite(*currentFile, inNumberFrames, ioData);
}
OSSpinLockUnlock(currentLock);
}
return AudioUnitProcess_orig(unit, ioActionFlags, inTimeStamp, inNumberFrames, ioData);
}
__attribute__((constructor))
static void initialize()
{
CTTelephonyCenterAddObserver(CTTelephonyCenterGetDefault(), NULL, CoreTelephonyNotificationCallback, NULL, NULL, CFNotificationSuspensionBehaviorHold);
MSHookFunction(AudioUnitProcess, AudioUnitProcess_hook, &AudioUnitProcess_orig);
}
A few words about what's going on. AudioUnitProcess function is used for processing audio streams in order to apply some effects, mix, convert etc. We are hooking AudioUnitProcess in order to access phone call's audio streams. While phone call is active these streams are being processed in various ways.
We are listening for CoreTelephony notifications in order to get phone call status changes. When we receive audio samples we need to determine where they come from - microphone or speaker. This is done using componentSubType field in AudioComponentDescription structure. Now, you might think, why don't we store AudioUnit objects so that we don't need to check componentSubType every time. I did that but it will break everything when you switch speaker on/off on iPhone 5 because AudioUnit objects will change, they are recreated. So, now we open audio files (one for microphone and one for speaker) and write samples in them, simple as that. When phone call ends we will receive appropriate CoreTelephony notification and close the files. We have two separate files with audio from microphone and speaker that we need to merge. This is what void Convert() is for. It's pretty simple if you know the API. I don't think I need to explain it, comments are enough.
About locks. There are many threads in mediaserverd. Audio processing and CoreTelephony notifications are on different threads so we need some kind synchronization. I chose spin locks because they are fast and because the chance of lock contention is small in our case. On iPhone 4S and even iPhone 5 all the work in AudioUnitProcess should be done as fast as possible otherwise you will hear hiccups from device speaker which obviously not good.
Yes. Audio Recorder by a developer named Limneos does that (and quite well). You can find it on Cydia. It can record any type of call on iPhone 5 and up without using any servers etc'. The call will be placed on the device in an Audio file. It also supports iPhone 4S but for speaker only.
This tweak is known to be the first tweak ever that managed to record both streams of audio without using any 3rd party severs, VOIP or something similar.
The developer placed beeps on the other side of the call to alert the person you are recording but those were removed too by hackers across the net. To answer your question, Yes, it's very much possible, and not just theoretically.
Further reading
https://stackoverflow.com/a/19413363/202451
http://forums.macrumors.com/showthread.php?t=1566350
https://github.com/nst/iOS-Runtime-Headers
The only solution I can think of is to use the Core Telephony framework, and more specifically the callEventHandler property, to intercept when a call is coming in, and then to use an AVAudioRecorder to record the voice of the person with the phone (and maybe a little of the person on the other line's voice). This is obviously not perfect, and would only work if your application is in the foreground at the time of the call, but it may be the best you can get. See more about finding out if there is an incoming phone call here: Can we fire an event when ever there is Incoming and Outgoing call in iphone?.
EDIT:
.h:
#import <AVFoundation/AVFoundation.h>
#import<CoreTelephony/CTCallCenter.h>
#import<CoreTelephony/CTCall.h>
#property (strong, nonatomic) AVAudioRecorder *audioRecorder;
ViewDidLoad:
NSArray *dirPaths;
NSString *docsDir;
dirPaths = NSSearchPathForDirectoriesInDomains(
NSDocumentDirectory, NSUserDomainMask, YES);
docsDir = dirPaths[0];
NSString *soundFilePath = [docsDir
stringByAppendingPathComponent:#"sound.caf"];
NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath];
NSDictionary *recordSettings = [NSDictionary
dictionaryWithObjectsAndKeys:
[NSNumber numberWithInt:AVAudioQualityMin],
AVEncoderAudioQualityKey,
[NSNumber numberWithInt:16],
AVEncoderBitRateKey,
[NSNumber numberWithInt: 2],
AVNumberOfChannelsKey,
[NSNumber numberWithFloat:44100.0],
AVSampleRateKey,
nil];
NSError *error = nil;
_audioRecorder = [[AVAudioRecorder alloc]
initWithURL:soundFileURL
settings:recordSettings
error:&error];
if (error)
{
NSLog(#"error: %#", [error localizedDescription]);
} else {
[_audioRecorder prepareToRecord];
}
CTCallCenter *callCenter = [[CTCallCenter alloc] init];
[callCenter setCallEventHandler:^(CTCall *call) {
if ([[call callState] isEqual:CTCallStateConnected]) {
[_audioRecorder record];
} else if ([[call callState] isEqual:CTCallStateDisconnected]) {
[_audioRecorder stop];
}
}];
AppDelegate.m:
- (void)applicationDidEnterBackground:(UIApplication *)application//Makes sure that the recording keeps happening even when app is in the background, though only can go for 10 minutes.
{
__block UIBackgroundTaskIdentifier task = 0;
task=[application beginBackgroundTaskWithExpirationHandler:^{
NSLog(#"Expiration handler called %f",[application backgroundTimeRemaining]);
[application endBackgroundTask:task];
task=UIBackgroundTaskInvalid;
}];
This is the first time using many of these features, so not sure if this is exactly right, but I think you get the idea. Untested, as I do not have access to the right tools at the moment. Compiled using these sources:
Recording voice in background using AVAudioRecorder
http://prassan-warrior.blogspot.com/2012/11/recording-audio-on-iphone-with.html
Can we fire an event when ever there is Incoming and Outgoing call in iphone?
Apple does not allow it and does not provide any API for it.
However, on a jailbroken device I'm sure it's possible. As a matter of fact, I think it's already done. I remember seeing an app when my phone was jailbroken that changed your voice and recorded the call - I remember it was a US company offering it only in the states. Unfortunately I don't remember the name...
I guess some hardware could solve this. Connected to the minijack-port; having earbuds and a microphone passing through a small recorder. This recorder can be very simple. While not in conversation the recorder could feed the phone with data/the recording (through the jack-cable). And with a simple start button (just like the volum controls on the earbuds) could be sufficient for timing the recording.
Some setups
http://www.danmccomb.com/posts/483/how-to-record-iphone-conversations-using-zoom-h4n/
http://forums.macrumors.com/showthread.php?t=346430