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I try to receive audio data from BlackHole2ch or other loop audio plugins on macOS.
I run this test code for macOS app. It callback with all 0 data.
#import "ViewController.h"
#implementation ViewController
#define DEVICE_UID "BlackHole2ch_UID"
static AudioQueueRef s_queue;
static AudioQueueBufferRef s_buffers[3];
- (void)viewDidLoad {
[super viewDidLoad];
[self audioInit];
[self audioStart];
}
- (void) audioInit
{
AudioStreamBasicDescription format;
format.mSampleRate = 44100;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
format.mBitsPerChannel = 16;
format.mChannelsPerFrame = 1;
format.mBytesPerFrame = 2 * format.mChannelsPerFrame;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
format.mReserved = 0;
OSStatus err = AudioQueueNewInput(&format, virtual_speaker_callback, (__bridge void *)(self),
CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &s_queue);
NSLog(#"AudioQueueNewInput err=%d AudioQueue=%p", err, s_queue);
CFStringRef deviceUID = CFStringCreateWithCString(kCFAllocatorDefault, DEVICE_UID, kCFStringEncodingMacRoman);
err = AudioQueueSetProperty(s_queue, kAudioQueueProperty_CurrentDevice, &deviceUID, sizeof(deviceUID));
NSLog(#"Set target device:%# err=%d", deviceUID, err);
for (int i = 0; i < NUM_BUFFERS; i++)
{
err = AudioQueueAllocateBuffer(s_queue, BUFFER_SIZE, &s_buffers[i]);
NSLog(#"AudioQueueAllocateBuffer err=%d", err);
err = AudioQueueEnqueueBuffer (s_queue, s_buffers[i], 0, NULL);
NSLog(#"AudioQueueEnqueueBuffer err=%d", err);
}
}
- (void) audioStart
{
OSStatus err = AudioQueueStart(s_queue, NULL);
NSLog(#"AudioQueueStart err=%d", err);
}
static void virtual_speaker_callback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, const AudioTimeStamp *inStartTime, UInt32 inNumberPacketDescriptions, const AudioStreamPacketDescription *inPacketDescs)
{
uint8_t *p = (uint8_t*)inBuffer->mAudioData;
int64_t sum = 0;
for(int i=0;i<inBuffer->mAudioDataBytesCapacity;++i) {
sum += p[i];
}
if (sum == 0) {
NSLog(#"virtual_speaker_callback all Zero ptr=%p data[]=%d %d ...", p, p[0], p[1]);
} else {
NSLog(#"virtual_speaker_callback valid data ptr=%p data[]=%d %d ...", p, p[0], p[1]);
}
OSStatus err = AudioQueueEnqueueBuffer(s_queue, inBuffer, 0, nil);
NSLog(#"AudioQueueEnqueueBuffer err=%d", err);
}
#end
I also try console app with same kind of code. It works with valid callback data.
I enabled "Audio Input" on Xcode capabilities settings.
MacOS:13.0.1 Ventura
Xcode:14.1 (14B47b)
I am trying to reduce the buffer size of the recorded Audio data (and later i want to convert it back and play using Audiounit), using AudioConverterFillComplexBuffer, the code i use is given below.
- (void)captureOutput:(AVCaptureOutput *)captureOutput
didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
fromConnection:(AVCaptureConnection *)connection
{
SInt16 buffer[length];
CMBlockBufferCopyDataBytes(blockBufferRef, 0, length, buffer);
NSData * data = [[NSData alloc] initWithBytes:buffer length:length];
if(!fConverter)
{
[self ConverterSetup:asbd1];
}
[self Convert:data];
}
-(void)ConverterSetup:(AudioStreamBasicDescription)sourceDesc
{
AudioStreamBasicDescription fOutputFormat = {0};
memset(&fOutputFormat, 0, sizeof(fOutputFormat));
fOutputFormat.mSampleRate = 32000;
fOutputFormat.mFormatID = kAudioFormatLinearPCM;
fOutputFormat.mFormatFlags = kAudioFormatFlagIsBigEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
fOutputFormat.mBytesPerPacket = fOutputFormat.mBytesPerFrame =4;//4 * sizeof(SInt16);
fOutputFormat.mFramesPerPacket = 1;
fOutputFormat.mChannelsPerFrame = 1;
fOutputFormat.mBitsPerChannel = 32;
OSStatus err = AudioConverterNew(&sourceDesc, &fOutputFormat,&fConverter); //converts mFormat -> mFileStreamFormat
if(err != noErr)
{
NSError *error = [NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
NSLog(#"Error = %#", error);
return ;
}
SInt32 channelMap[] = { 0, 0 };
err = AudioConverterSetProperty(fConverter, kAudioConverterChannelMap, 2*sizeof(SInt32), channelMap);
UInt32 quality = kAudioConverterQuality_Medium;
err = AudioConverterSetProperty(fConverter,
kAudioConverterSampleRateConverterQuality,
sizeof(UInt32),
&quality);
}
-(void)Convert:(NSData*)data
{
AudioBufferList inBufferList_new;
inBufferList_new.mNumberBuffers = 1;
inBufferList_new.mBuffers[0].mNumberChannels = 1;
inBufferList_new.mBuffers[0].mData = (void *)data.bytes;
inBufferList_new.mBuffers[0].mDataByteSize = [data length];
char szBuf[1024];
int nSize = sizeof(szBuf);
AudioBufferList fAudioOutputBuffer;
fAudioOutputBuffer.mNumberBuffers = 1;
fAudioOutputBuffer.mBuffers[0].mNumberChannels = 1;
fAudioOutputBuffer.mBuffers[0].mDataByteSize = nSize;
fAudioOutputBuffer.mBuffers[0].mData = szBuf;
UInt32 outputDataPacketSize = nSize;
UInt numPackets = nSize;//inBufferList_new.mBuffers[0].mDataByteSize;
OSStatus err = AudioConverterFillComplexBuffer(fConverter, ConverterProc, &inBufferList_new, &numPackets, &fAudioOutputBuffer, NULL);
if(err != noErr)
{
NSError *error = [NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
NSLog(#"%#", error);
}
else
{
NSData *data=[NSData dataWithBytes:(UInt8*)fAudioOutputBuffer.mBuffers[0].mData length:fAudioOutputBuffer.mBuffers[0].mDataByteSize];
NSLog(#"Converted data length = %d", [data length]);
}
}
OSStatus ConverterProc(AudioConverterRef inAudioConverter,
UInt32* ioNumberDataPackets,
AudioBufferList* ioData,
AudioStreamPacketDescription** outDataPacketDescription,
void* inUserData)
{
OSStatus err = kAudioUnitErr_InvalidPropertyValue;
AudioBufferList bufferList = *(AudioBufferList*)inUserData;
ioData->mBuffers[0].mNumberChannels = 1;
ioData->mBuffers[0].mData = bufferList.mBuffers[0].mData;
ioData->mBuffers[0].mDataByteSize = bufferList.mBuffers[0].mDataByteSize;
return err;
}
The error i am getting is Error Domain=NSOSStatusErrorDomain Code=-10851 "(null)".
Can anyone figure out the problem with this code.
https://android.googlesource.com/platform/external/mpg123/+/3d540f5de5b3a28ce6ad855cef7d9d9a44242c07/src/output/coreaudio.c
can see the code :
if(ao->channels == 1) {
SInt32 channelMap[2] = { 0, 0 };
if(AudioConverterSetProperty(ca->converter, kAudioConverterChannelMap, sizeof(channelMap), channelMap)) {
error("AudioConverterSetProperty(kAudioConverterChannelMap) failed");
return(-1);
}
}
I'm trying to record sound from the microphone and play it back in real time on OS X. Eventually it will be streamed over the network, but for now I'm just trying to achieve local recording/playback.
I'm able to record sound and write to a file, which I could do with both AVCaptureSession and AVAudioRecorder. However, I'm not sure how to play back the audio as I record it. Using AVCaptureAudioDataOutput works:
self.captureSession = [[AVCaptureSession alloc] init];
AVCaptureDevice *audioCaptureDevice = [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeAudio];
NSError *error = nil;
AVCaptureDeviceInput *audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioCaptureDevice error:&error];
AVCaptureAudioDataOutput *audioDataOutput = [[AVCaptureAudioDataOutput alloc] init];
self.serialQueue = dispatch_queue_create("audioQueue", NULL);
[audioDataOutput setSampleBufferDelegate:self queue:self.serialQueue];
if (audioInput && [self.captureSession canAddInput:audioInput] && [self.captureSession canAddOutput:audioDataOutput]) {
[self.captureSession addInput:audioInput];
[self.captureSession addOutput:audioDataOutput];
[self.captureSession startRunning];
// Stop after arbitrary time
double delayInSeconds = 4.0;
dispatch_time_t popTime = dispatch_time(DISPATCH_TIME_NOW, (int64_t)(delayInSeconds * NSEC_PER_SEC));
dispatch_after(popTime, dispatch_get_main_queue(), ^(void){
[self.captureSession stopRunning];
});
} else {
NSLog(#"Couldn't add them; error = %#",error);
}
...but I'm not sure how to implement the callback:
- (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection
{
?
}
I've tried getting the data out of the sampleBuffer and playing it using AVAudioPlayer by copying the code from this SO answer, but that code crashes on the appendBytes:length: method.
AudioBufferList audioBufferList;
NSMutableData *data= [NSMutableData data];
CMBlockBufferRef blockBuffer;
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, NULL, &audioBufferList, sizeof(audioBufferList), NULL, NULL, 0, &blockBuffer);
for( int y=0; y< audioBufferList.mNumberBuffers; y++ ){
AudioBuffer audioBuffer = audioBufferList.mBuffers[y];
Float32 *frame = (Float32*)audioBuffer.mData;
NSLog(#"Length = %i",audioBuffer.mDataByteSize);
[data appendBytes:frame length:audioBuffer.mDataByteSize]; // Crashes here
}
CFRelease(blockBuffer);
NSError *playerError;
AVAudioPlayer *player = [[AVAudioPlayer alloc] initWithData:data error:&playerError];
if(player && !playerError) {
NSLog(#"Player was valid");
[player play];
} else {
NSLog(#"Error = %#",playerError);
}
Edit The CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer method returns an OSStatus code of -12737, which according to the documentation is kCMSampleBufferError_ArrayTooSmall
Edit2: Based on this mailing list response, I passed a size_t out parameter as the second parameter to ...GetAudioBufferList.... This returned 40. Right now I'm just passing in 40 as a hard-coded value, which seems to work (the OSStatus return value is 0, atleast).
Now the player initWithData:error: method gives the error:
Error Domain=NSOSStatusErrorDomain Code=1954115647 "The operation couldn’t be completed. (OSStatus error 1954115647.)" which I'm looking into.
I've done iOS programming for a long time, but I haven't used AVFoundation, CoreAudio, etc until now. It looks like there are a dozen ways to accomplish the same thing, depending on how low or high level you want to be, so any high level overviews or framework recommendations are appreciated.
Appendix
Recording to a file
Recording to a file using AVCaptureSession:
- (void)applicationDidFinishLaunching:(NSNotification *)aNotification
{
[[NSNotificationCenter defaultCenter] addObserver:self selector:#selector(captureSessionStartedNotification:) name:AVCaptureSessionDidStartRunningNotification object:nil];
self.captureSession = [[AVCaptureSession alloc] init];
AVCaptureDevice *audioCaptureDevice = [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeAudio];
NSError *error = nil;
AVCaptureDeviceInput *audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioCaptureDevice error:&error];
AVCaptureAudioFileOutput *audioOutput = [[AVCaptureAudioFileOutput alloc] init];
if (audioInput && [self.captureSession canAddInput:audioInput] && [self.captureSession canAddOutput:audioOutput]) {
NSLog(#"Can add the inputs and outputs");
[self.captureSession addInput:audioInput];
[self.captureSession addOutput:audioOutput];
[self.captureSession startRunning];
double delayInSeconds = 5.0;
dispatch_time_t popTime = dispatch_time(DISPATCH_TIME_NOW, (int64_t)(delayInSeconds * NSEC_PER_SEC));
dispatch_after(popTime, dispatch_get_main_queue(), ^(void){
[self.captureSession stopRunning];
});
}
else {
NSLog(#"Error was = %#",error);
}
}
- (void)captureSessionStartedNotification:(NSNotification *)notification
{
AVCaptureSession *session = notification.object;
id audioOutput = session.outputs[0];
NSLog(#"Capture session started; notification = %#",notification);
NSLog(#"Notification audio output = %#",audioOutput);
[audioOutput startRecordingToOutputFileURL:[[self class] outputURL] outputFileType:AVFileTypeAppleM4A recordingDelegate:self];
}
+ (NSURL *)outputURL
{
NSArray *searchPaths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
NSString *documentPath = [searchPaths objectAtIndex:0];
NSString *filePath = [documentPath stringByAppendingPathComponent:#"z1.alac"];
return [NSURL fileURLWithPath:filePath];
}
Recording to a file using AVAudioRecorder:
NSDictionary *recordSettings = [NSDictionary
dictionaryWithObjectsAndKeys:
[NSNumber numberWithInt:AVAudioQualityMin],
AVEncoderAudioQualityKey,
[NSNumber numberWithInt:16],
AVEncoderBitRateKey,
[NSNumber numberWithInt: 2],
AVNumberOfChannelsKey,
[NSNumber numberWithFloat:44100.0],
AVSampleRateKey,
#(kAudioFormatAppleLossless),
AVFormatIDKey,
nil];
NSError *recorderError;
self.recorder = [[AVAudioRecorder alloc] initWithURL:[[self class] outputURL] settings:recordSettings error:&recorderError];
self.recorder.delegate = self;
if (self.recorder && !recorderError) {
NSLog(#"Success!");
[self.recorder recordForDuration:10];
} else {
NSLog(#"Failure, recorder = %#",self.recorder);
NSLog(#"Error = %#",recorderError);
}
Ok, I ended up working at a lower level than AVFoundation -- not sure if that was necessary. I read up to Chapter 5 of Learning Core Audio and went with an implementation using Audio Queues. This code is translated from being used for recording to a file/playing back a file, so there are surely some unnecessary bits I've accidentally left in. Additionally, I'm not actually re-enqueuing buffers onto the Output Queue (I should be), but just as a proof of concept this works. The only file is listed here, and is also on Github.
//
// main.m
// Recorder
//
// Created by Maximilian Tagher on 8/7/13.
// Copyright (c) 2013 Tagher. All rights reserved.
//
#import <Foundation/Foundation.h>
#import <AudioToolbox/AudioToolbox.h>
#define kNumberRecordBuffers 3
//#define kNumberPlaybackBuffers 3
#define kPlaybackFileLocation CFSTR("/Users/Max/Music/iTunes/iTunes Media/Music/Taylor Swift/Red/02 Red.m4a")
#pragma mark - User Data Struct
// listing 4.3
struct MyRecorder;
typedef struct MyPlayer {
AudioQueueRef playerQueue;
SInt64 packetPosition;
UInt32 numPacketsToRead;
AudioStreamPacketDescription *packetDescs;
Boolean isDone;
struct MyRecorder *recorder;
} MyPlayer;
typedef struct MyRecorder {
AudioQueueRef recordQueue;
SInt64 recordPacket;
Boolean running;
MyPlayer *player;
} MyRecorder;
#pragma mark - Utility functions
// Listing 4.2
static void CheckError(OSStatus error, const char *operation) {
if (error == noErr) return;
char errorString[20];
// See if it appears to be a 4-char-code
*(UInt32 *)(errorString + 1) = CFSwapInt32HostToBig(error);
if (isprint(errorString[1]) && isprint(errorString[2])
&& isprint(errorString[3]) && isprint(errorString[4])) {
errorString[0] = errorString[5] = '\'';
errorString[6] = '\0';
} else {
// No, format it as an integer
NSLog(#"Was integer");
sprintf(errorString, "%d",(int)error);
}
fprintf(stderr, "Error: %s (%s)\n",operation,errorString);
exit(1);
}
OSStatus MyGetDefaultInputDeviceSampleRate(Float64 *outSampleRate)
{
OSStatus error;
AudioDeviceID deviceID = 0;
AudioObjectPropertyAddress propertyAddress;
UInt32 propertySize;
propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = 0;
propertySize = sizeof(AudioDeviceID);
error = AudioHardwareServiceGetPropertyData(kAudioObjectSystemObject,
&propertyAddress, 0, NULL,
&propertySize,
&deviceID);
if (error) return error;
propertyAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = 0;
propertySize = sizeof(Float64);
error = AudioHardwareServiceGetPropertyData(deviceID,
&propertyAddress, 0, NULL,
&propertySize,
outSampleRate);
return error;
}
// Recorder
static void MyCopyEncoderCookieToFile(AudioQueueRef queue, AudioFileID theFile)
{
OSStatus error;
UInt32 propertySize;
error = AudioQueueGetPropertySize(queue, kAudioConverterCompressionMagicCookie, &propertySize);
if (error == noErr && propertySize > 0) {
Byte *magicCookie = (Byte *)malloc(propertySize);
CheckError(AudioQueueGetProperty(queue, kAudioQueueProperty_MagicCookie, magicCookie, &propertySize), "Couldn't get audio queue's magic cookie");
CheckError(AudioFileSetProperty(theFile, kAudioFilePropertyMagicCookieData, propertySize, magicCookie), "Couldn't set audio file's magic cookie");
free(magicCookie);
}
}
// Player
static void MyCopyEncoderCookieToQueue(AudioFileID theFile, AudioQueueRef queue)
{
UInt32 propertySize;
// Just check for presence of cookie
OSStatus result = AudioFileGetProperty(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 file failed");
free(magicCookie);
}
}
static int MyComputeRecordBufferSize(const AudioStreamBasicDescription *format, AudioQueueRef queue, float seconds)
{
int packets, frames, bytes;
frames = (int)ceil(seconds * format->mSampleRate);
if (format->mBytesPerFrame > 0) { // Not variable
bytes = frames * format->mBytesPerFrame;
} else { // variable bytes per frame
UInt32 maxPacketSize;
if (format->mBytesPerPacket > 0) {
// Constant packet size
maxPacketSize = format->mBytesPerPacket;
} else {
// Get the largest single packet size possible
UInt32 propertySize = sizeof(maxPacketSize);
CheckError(AudioQueueGetProperty(queue, kAudioConverterPropertyMaximumOutputPacketSize, &maxPacketSize, &propertySize), "Couldn't get queue's maximum output packet size");
}
if (format->mFramesPerPacket > 0) {
packets = frames / format->mFramesPerPacket;
} else {
// Worst case scenario: 1 frame in a packet
packets = frames;
}
// Sanity check
if (packets == 0) {
packets = 1;
}
bytes = packets * maxPacketSize;
}
return bytes;
}
void CalculateBytesForPlaythrough(AudioQueueRef queue,
AudioStreamBasicDescription inDesc,
Float64 inSeconds,
UInt32 *outBufferSize,
UInt32 *outNumPackets)
{
UInt32 maxPacketSize;
UInt32 propSize = sizeof(maxPacketSize);
CheckError(AudioQueueGetProperty(queue,
kAudioQueueProperty_MaximumOutputPacketSize,
&maxPacketSize, &propSize), "Couldn't get file's max packet size");
static const int maxBufferSize = 0x10000;
static const int minBufferSize = 0x4000;
if (inDesc.mFramesPerPacket) {
Float64 numPacketsForTime = inDesc.mSampleRate / inDesc.mFramesPerPacket * inSeconds;
*outBufferSize = numPacketsForTime * maxPacketSize;
} else {
*outBufferSize = maxBufferSize > maxPacketSize ? maxBufferSize : maxPacketSize;
}
if (*outBufferSize > maxBufferSize &&
*outBufferSize > maxPacketSize) {
*outBufferSize = maxBufferSize;
} else {
if (*outBufferSize < minBufferSize) {
*outBufferSize = minBufferSize;
}
}
*outNumPackets = *outBufferSize / maxPacketSize;
}
#pragma mark - Record callback function
static void MyAQInputCallback(void *inUserData,
AudioQueueRef inQueue,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc)
{
// NSLog(#"Input callback");
// NSLog(#"Input thread = %#",[NSThread currentThread]);
MyRecorder *recorder = (MyRecorder *)inUserData;
MyPlayer *player = recorder->player;
if (inNumPackets > 0) {
// Enqueue on the output Queue!
AudioQueueBufferRef outputBuffer;
CheckError(AudioQueueAllocateBuffer(player->playerQueue, inBuffer->mAudioDataBytesCapacity, &outputBuffer), "Input callback failed to allocate new output buffer");
memcpy(outputBuffer->mAudioData, inBuffer->mAudioData, inBuffer->mAudioDataByteSize);
outputBuffer->mAudioDataByteSize = inBuffer->mAudioDataByteSize;
// [NSData dataWithBytes:inBuffer->mAudioData length:inBuffer->mAudioDataByteSize];
// Assuming LPCM so no packet descriptions
CheckError(AudioQueueEnqueueBuffer(player->playerQueue, outputBuffer, 0, NULL), "Enqueing the buffer in input callback failed");
recorder->recordPacket += inNumPackets;
}
if (recorder->running) {
CheckError(AudioQueueEnqueueBuffer(inQueue, inBuffer, 0, NULL), "AudioQueueEnqueueBuffer failed");
}
}
static void MyAQOutputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inCompleteAQBuffer)
{
// NSLog(#"Output thread = %#",[NSThread currentThread]);
// NSLog(#"Output callback");
MyPlayer *aqp = (MyPlayer *)inUserData;
MyRecorder *recorder = aqp->recorder;
if (aqp->isDone) return;
}
int main(int argc, const char * argv[])
{
#autoreleasepool {
MyRecorder recorder = {0};
MyPlayer player = {0};
recorder.player = &player;
player.recorder = &recorder;
AudioStreamBasicDescription recordFormat;
memset(&recordFormat, 0, sizeof(recordFormat));
recordFormat.mFormatID = kAudioFormatLinearPCM;
recordFormat.mChannelsPerFrame = 2; //stereo
// Begin my changes to make LPCM work
recordFormat.mBitsPerChannel = 16;
// Haven't checked if each of these flags is necessary, this is just what Chapter 2 used for LPCM.
recordFormat.mFormatFlags = kAudioFormatFlagIsBigEndian | kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
// end my changes
MyGetDefaultInputDeviceSampleRate(&recordFormat.mSampleRate);
UInt32 propSize = sizeof(recordFormat);
CheckError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
0,
NULL,
&propSize,
&recordFormat), "AudioFormatGetProperty failed");
AudioQueueRef queue = {0};
CheckError(AudioQueueNewInput(&recordFormat, MyAQInputCallback, &recorder, NULL, NULL, 0, &queue), "AudioQueueNewInput failed");
recorder.recordQueue = queue;
// Fills in ABSD a little more
UInt32 size = sizeof(recordFormat);
CheckError(AudioQueueGetProperty(queue,
kAudioConverterCurrentOutputStreamDescription,
&recordFormat,
&size), "Couldn't get queue's format");
// MyCopyEncoderCookieToFile(queue, recorder.recordFile);
int bufferByteSize = MyComputeRecordBufferSize(&recordFormat,queue,0.5);
NSLog(#"%d",__LINE__);
// Create and Enqueue buffers
int bufferIndex;
for (bufferIndex = 0;
bufferIndex < kNumberRecordBuffers;
++bufferIndex) {
AudioQueueBufferRef buffer;
CheckError(AudioQueueAllocateBuffer(queue,
bufferByteSize,
&buffer), "AudioQueueBufferRef failed");
CheckError(AudioQueueEnqueueBuffer(queue, buffer, 0, NULL), "AudioQueueEnqueueBuffer failed");
}
// PLAYBACK SETUP
AudioQueueRef playbackQueue;
CheckError(AudioQueueNewOutput(&recordFormat,
MyAQOutputCallback,
&player, NULL, NULL, 0,
&playbackQueue), "AudioOutputNewQueue failed");
player.playerQueue = playbackQueue;
UInt32 playBufferByteSize;
CalculateBytesForPlaythrough(queue, recordFormat, 0.1, &playBufferByteSize, &player.numPacketsToRead);
bool isFormatVBR = (recordFormat.mBytesPerPacket == 0
|| recordFormat.mFramesPerPacket == 0);
if (isFormatVBR) {
NSLog(#"Not supporting VBR");
player.packetDescs = (AudioStreamPacketDescription*) malloc(sizeof(AudioStreamPacketDescription) * player.numPacketsToRead);
} else {
player.packetDescs = NULL;
}
// END PLAYBACK
recorder.running = TRUE;
player.isDone = false;
CheckError(AudioQueueStart(playbackQueue, NULL), "AudioQueueStart failed");
CheckError(AudioQueueStart(queue, NULL), "AudioQueueStart failed");
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 10, TRUE);
printf("Playing through, press <return> to stop:\n");
getchar();
printf("* done *\n");
recorder.running = FALSE;
player.isDone = true;
CheckError(AudioQueueStop(playbackQueue, false), "Failed to stop playback queue");
CheckError(AudioQueueStop(queue, TRUE), "AudioQueueStop failed");
AudioQueueDispose(playbackQueue, FALSE);
AudioQueueDispose(queue, TRUE);
}
return 0;
}
guys! I have a trouble with using remote IO to playback a stream audio.I verified the PCM frame data before I put it in,it's correct.So I'm confused.Could you help me? Thanks a lot!
Below is my codes.
-
(void)initializeAudioPlay
{
OSStatus status;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// Get audio units
status = AudioComponentInstanceNew(inputComponent, &audioPlayUnit);
[self checkStatus:status];
// Enable IO for playback
UInt32 flag = 1;
//kAUVoiceIOProperty_VoiceProcessingEnableAGC
status = AudioUnitSetProperty(audioPlayUnit, kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input, kOutputBus, &flag, sizeof(flag));
[self checkStatus:status];
// Describe format
AudioStreamBasicDescription audioFormat;
memset(&audioFormat, 0, sizeof(audioFormat));
audioFormat.mSampleRate = 8000;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagsCanonical;//kAudioFormatFlagIsNonInterleaved | kAudioFormatFlagIsSignedInteger;
/*kAudioFormatFlagsCanonical
| (kAudioUnitSampleFractionBits << kLinearPCMFormatFlagsSampleFractionShift)*/
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerFrame = (audioFormat.mBitsPerChannel/8) * audioFormat.mChannelsPerFrame;
audioFormat.mBytesPerPacket = audioFormat.mBytesPerFrame;
// Apply format
status = AudioUnitSetProperty(audioPlayUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
[self checkStatus:status];
float value = (float)10 / 255.0;
AudioUnitSetParameter(audioPlayUnit, kAudioUnitParameterUnit_LinearGain, kAudioUnitScope_Input, 0, value, 0);
AudioChannelLayout new_layout;
new_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
AudioUnitSetProperty( audioPlayUnit,
kAudioUnitProperty_AudioChannelLayout,
kAudioUnitScope_Global,
0, &new_layout, sizeof(new_layout) );
UInt32 bypassEffect = kAudioUnitProperty_RenderQuality;
status = AudioUnitSetProperty(audioPlayUnit,
kAudioUnitProperty_RenderQuality,
kAudioUnitScope_Global,
0,
&bypassEffect,
sizeof(bypassEffect));
[self checkStatus:status];
// Set output callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = playCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioPlayUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
[self checkStatus:status];
flag = 0;
// Initialize
status = AudioUnitInitialize(audioPlayUnit);
[self checkStatus:status];
DGLog(#"audio play unit initialize = %d", status);
circularBuf = [[CircularBuf alloc] initWithBufLen:kBufferLength];
/*
AudioSessionInitialize(NULL, NULL, NULL, NULL);
Float64 rate =32000.0;
AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareSampleRate, sizeof(rate), &rate);
Float32 volume=20.0;
UInt32 size = sizeof(Float32);
AudioSessionSetProperty(
kAudioSessionProperty_PreferredHardwareIOBufferDuration,
&size, &volume);
//float aBufferLength = 0.185759637188209;
//AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(aBufferLength), &aBufferLength);
AudioSessionSetActive(YES);
*/
AudioSessionInitialize(NULL, NULL, NULL, nil);
AudioSessionSetActive(true);
UInt32 sessionCategory = kAudioSessionCategory_MediaPlayback ;
/* for Iphone we need to do this to route the audio to speaker */
status= AudioSessionSetProperty (
kAudioSessionProperty_AudioCategory,
sizeof (sessionCategory),
&sessionCategory
);
//NSLog(#"Error: %d", status);
//
// UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker;
// status = AudioSessionSetProperty (
// kAudioSessionProperty_OverrideAudioRoute,
// sizeof (audioRouteOverride),
// &audioRouteOverride);
UInt32 audioMixed = 1;
status = AudioSessionSetProperty (
kAudioSessionProperty_OverrideCategoryMixWithOthers,
sizeof (audioMixed),
&audioMixed);
}
- (void)processAudio:(AudioBuffer *)buffer
{
short pcmTemp[160];
unsigned char * amrBuffer=NULL;
AudioUnitSampleType sample;
int i = 0;
int j = 0;
if ([circularBuf isReadTwoRegion]) {
amrBuffer = [circularBuf ReadData];
} else {
amrBuffer = [circularBuf ReadData];
i = [circularBuf ReadPos];
}
j = i + circularBuf.Length;
if (j - i >= 320) {
memcpy((void*)pcmTemp, (void*)amrBuffer, 320);
for(i=0; i<160; i++)
{
sample = 3.162277*pcmTemp[i];//10db
if(sample > 32767)sample = 32767;
else if(sample < -32768)sample = -32768;
buffData[i] = sample;
}
memcpy(buffer->mData, buffData, buffer->mDataByteSize);
[circularBuf AdvanceReadPos:320];
}
else
{
memset(buffer->mData, 0, buffer->mDataByteSize);
}
}
/**
This callback is called when the audioUnit needs new data to play through the
speakers. If you don't have any, just don't write anything in the buffers
*/
static OSStatus playCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
AudioPlay *audioPlay = (AudioPlay *)inRefCon;
for ( int i=0; i < ioData->mNumberBuffers; i++ ) {
memset(ioData->mBuffers[i].mData, 0, ioData->mBuffers[i].mDataByteSize);
}
ioData->mBuffers[0].mNumberChannels = 1;
[audioPlay processAudio:&ioData->mBuffers[0]];
return noErr;
}
I wanna resize a picture by using the ffmpeg's func--->sws_scale().
Is there any one knows how to do it?
Do you have the source code for this function?
First you need to create a SwsContext (you need to do this only once) :
struct SwsContext *resize;
resize = sws_getContext(width1, height1, AV_PIX_FMT_YUV420P, width2, height2, PIX_FMT_RGB24, SWS_BICUBIC, NULL, NULL, NULL);
You need two frames for conversion, frame1 is the original frame, you need to explicitly allocate frame2 :
AVFrame* frame1 = avcodec_alloc_frame(); // this is your original frame
AVFrame* frame2 = avcodec_alloc_frame();
int num_bytes = avpicture_get_size(AV_PIX_FMT_RGB24, width2, height2);
uint8_t* frame2_buffer = (uint8_t *)av_malloc(num_bytes*sizeof(uint8_t));
avpicture_fill((AVPicture*)frame2, frame2_buffer, AV_PIX_FMT_RGB24, width2, height2);
You may use this part inside a loop if you need to resize each frame you receive :
// frame1 should be filled by now (eg using avcodec_decode_video)
sws_scale(resize, frame1->data, frame1->linesize, 0, height1, frame2->data, frame2->linesize);
Note that I also changed pixel format, but you can use the same pixel format for both frames
Runnable example in FFmpeg 2.8
Basically using arash's method, but runnable so you can try it out.
Generate one short video procedurally, and then convert it to 3 different sizes.
ffmpeg_encoder_init_frame and ffmpeg_encoder_scale are the key methods.
Source:
#include <libavcodec/avcodec.h>
#include <libavutil/imgutils.h>
#include <libavutil/opt.h>
#include <libswscale/swscale.h>
static AVCodecContext *c = NULL;
static AVFrame *frame;
static AVFrame *frame2;
static AVPacket pkt;
static FILE *file;
static struct SwsContext *sws_context = NULL;
static void ffmpeg_encoder_init_frame(AVFrame **framep, int width, int height) {
int ret;
AVFrame *frame;
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = width;
frame->height = height;
ret = av_image_alloc(frame->data, frame->linesize, frame->width, frame->height, frame->format, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
*framep = frame;
}
static void ffmpeg_encoder_scale(uint8_t *rgb) {
sws_context = sws_getCachedContext(sws_context,
frame->width, frame->height, AV_PIX_FMT_YUV420P,
frame2->width, frame2->height, AV_PIX_FMT_YUV420P,
SWS_BICUBIC, NULL, NULL, NULL);
sws_scale(sws_context, (const uint8_t * const *)frame->data, frame->linesize, 0,
frame->height, frame2->data, frame2->linesize);
}
static void ffmpeg_encoder_set_frame_yuv_from_rgb(uint8_t *rgb) {
const int in_linesize[1] = { 3 * frame->width };
sws_context = sws_getCachedContext(sws_context,
frame->width, frame->height, AV_PIX_FMT_RGB24,
frame->width, frame->height, AV_PIX_FMT_YUV420P,
0, NULL, NULL, NULL);
sws_scale(sws_context, (const uint8_t * const *)&rgb, in_linesize, 0,
frame->height, frame->data, frame->linesize);
}
void generate_rgb(int width, int height, int pts, uint8_t **rgbp) {
int x, y, cur;
uint8_t *rgb = *rgbp;
rgb = realloc(rgb, 3 * sizeof(uint8_t) * height * width);
for (y = 0; y < height; y++) {
for (x = 0; x < width; x++) {
cur = 3 * (y * width + x);
rgb[cur + 0] = 0;
rgb[cur + 1] = 0;
rgb[cur + 2] = 0;
if ((frame->pts / 25) % 2 == 0) {
if (y < height / 2) {
if (x < width / 2) {
/* Black. */
} else {
rgb[cur + 0] = 255;
}
} else {
if (x < width / 2) {
rgb[cur + 1] = 255;
} else {
rgb[cur + 2] = 255;
}
}
} else {
if (y < height / 2) {
rgb[cur + 0] = 255;
if (x < width / 2) {
rgb[cur + 1] = 255;
} else {
rgb[cur + 2] = 255;
}
} else {
if (x < width / 2) {
rgb[cur + 1] = 255;
rgb[cur + 2] = 255;
} else {
rgb[cur + 0] = 255;
rgb[cur + 1] = 255;
rgb[cur + 2] = 255;
}
}
}
}
}
*rgbp = rgb;
}
void ffmpeg_encoder_start(const char *filename, int codec_id, int fps, int width, int height, float factor) {
AVCodec *codec;
int ret;
int width2 = width * factor;
int height2 = height * factor;
avcodec_register_all();
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
c->bit_rate = 400000;
c->width = width2;
c->height = height2;
c->time_base.num = 1;
c->time_base.den = fps;
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
file = fopen(filename, "wb");
if (!file) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
ffmpeg_encoder_init_frame(&frame, width, height);
ffmpeg_encoder_init_frame(&frame2, width2, height2);
}
void ffmpeg_encoder_finish(void) {
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
int got_output, ret;
do {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, file);
av_packet_unref(&pkt);
}
} while (got_output);
fwrite(endcode, 1, sizeof(endcode), file);
fclose(file);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
av_freep(&frame2->data[0]);
av_frame_free(&frame2);
}
void ffmpeg_encoder_encode_frame(uint8_t *rgb) {
int ret, got_output;
ffmpeg_encoder_set_frame_yuv_from_rgb(rgb);
ffmpeg_encoder_scale(rgb);
frame2->pts = frame->pts;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
ret = avcodec_encode_video2(c, &pkt, frame2, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, file);
av_packet_unref(&pkt);
}
}
static void encode_example(float factor) {
char filename[255];
int pts;
int width = 320;
int height = 240;
uint8_t *rgb = NULL;
snprintf(filename, 255, "tmp." __FILE__ ".%.2f.h264", factor);
ffmpeg_encoder_start(filename, AV_CODEC_ID_H264, 25, width, height, factor);
for (pts = 0; pts < 100; pts++) {
frame->pts = pts;
generate_rgb(width, height, pts, &rgb);
ffmpeg_encoder_encode_frame(rgb);
}
ffmpeg_encoder_finish();
free(rgb);
}
int main(void) {
encode_example(0.5);
encode_example(1.0);
encode_example(2.0);
return EXIT_SUCCESS;
}
Run with:
gcc main.c -lavformat -lavcodec -lswresample -lswscale -lavutil -lx264
./a.out
ffplay tmp.main.c.0.50.h264
ffplay tmp.main.c.1.00.h264
ffplay tmp.main.c.2.00.h264
Tested on ffmpeg 3.4.11, Ubuntu 16.04. Source on GitHub. Adapted from doc/examples/decoding_encoding.c.