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add vizualizer for player buffers

This commit is contained in:
Николай Ашанин 2016-02-08 00:21:29 +03:00
parent a84db95be6
commit dfe6c73621
5 changed files with 369 additions and 0 deletions
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8
StreamingKit/StreamingKit.xcodeproj/project.pbxproj
View File

@ -15,6 +15,8 @@
5B949CD71A1140E4005675A0 /* STKHTTPDataSource.h in Headers */ = {isa = PBXBuildFile; fileRef = A1E7C4FB188D5E550010896F /* STKHTTPDataSource.h */; settings = {ATTRIBUTES = (Public, ); }; };
5B949CD81A1140E4005675A0 /* STKLocalFileDataSource.h in Headers */ = {isa = PBXBuildFile; fileRef = A1E7C4FD188D5E550010896F /* STKLocalFileDataSource.h */; settings = {ATTRIBUTES = (Public, ); }; };
5B949CD91A1140E4005675A0 /* STKQueueEntry.h in Headers */ = {isa = PBXBuildFile; fileRef = A1BF65D0189A6582004DD08C /* STKQueueEntry.h */; settings = {ATTRIBUTES = (Public, ); }; };
958A8E291C67E9B700AA861D /* AEFloatConverter.h in Headers */ = {isa = PBXBuildFile; fileRef = 958A8E271C67E9B700AA861D /* AEFloatConverter.h */; };
958A8E2A1C67E9B700AA861D /* AEFloatConverter.m in Sources */ = {isa = PBXBuildFile; fileRef = 958A8E281C67E9B700AA861D /* AEFloatConverter.m */; };
A1A4996B189E744400E2A2E2 /* Cocoa.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = A1A4996A189E744400E2A2E2 /* Cocoa.framework */; };
A1A49975189E744500E2A2E2 /* StreamingKitMac.m in Sources */ = {isa = PBXBuildFile; fileRef = A1A49974189E744500E2A2E2 /* StreamingKitMac.m */; };
A1A4997B189E744500E2A2E2 /* XCTest.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = A1E7C4D9188D57F60010896F /* XCTest.framework */; };
@ -91,6 +93,8 @@
/* End PBXCopyFilesBuildPhase section */
/* Begin PBXFileReference section */
958A8E271C67E9B700AA861D /* AEFloatConverter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AEFloatConverter.h; sourceTree = "<group>"; };
958A8E281C67E9B700AA861D /* AEFloatConverter.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = AEFloatConverter.m; sourceTree = "<group>"; };
A1A49969189E744400E2A2E2 /* libStreamingKitMac.a */ = {isa = PBXFileReference; explicitFileType = archive.ar; includeInIndex = 0; path = libStreamingKitMac.a; sourceTree = BUILT_PRODUCTS_DIR; };
A1A4996A189E744400E2A2E2 /* Cocoa.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Cocoa.framework; path = Library/Frameworks/Cocoa.framework; sourceTree = DEVELOPER_DIR; };
A1A4996D189E744500E2A2E2 /* Foundation.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Foundation.framework; path = Library/Frameworks/Foundation.framework; sourceTree = SDKROOT; };
@ -270,6 +274,8 @@
A1E7C4CD188D57F50010896F /* StreamingKit */ = {
isa = PBXGroup;
children = (
958A8E271C67E9B700AA861D /* AEFloatConverter.h */,
958A8E281C67E9B700AA861D /* AEFloatConverter.m */,
A1E7C4F1188D5E550010896F /* STKAudioPlayer.h */,
A1E7C4F2188D5E550010896F /* STKAudioPlayer.m */,
A1E7C4F3188D5E550010896F /* STKAutoRecoveringHTTPDataSource.h */,
@ -331,6 +337,7 @@
5B949CD41A1140E4005675A0 /* STKCoreFoundationDataSource.h in Headers */,
5B949CD51A1140E4005675A0 /* STKDataSource.h in Headers */,
5B949CD61A1140E4005675A0 /* STKDataSourceWrapper.h in Headers */,
958A8E291C67E9B700AA861D /* AEFloatConverter.h in Headers */,
5B949CD71A1140E4005675A0 /* STKHTTPDataSource.h in Headers */,
5B949CD81A1140E4005675A0 /* STKLocalFileDataSource.h in Headers */,
5B949CD91A1140E4005675A0 /* STKQueueEntry.h in Headers */,
@ -525,6 +532,7 @@
A1E7C505188D5E550010896F /* STKLocalFileDataSource.m in Sources */,
A1BF65D2189A6582004DD08C /* STKQueueEntry.m in Sources */,
A1E7C504188D5E550010896F /* STKHTTPDataSource.m in Sources */,
958A8E2A1C67E9B700AA861D /* AEFloatConverter.m in Sources */,
A1E7C503188D5E550010896F /* STKDataSourceWrapper.m in Sources */,
A1E7C502188D5E550010896F /* STKDataSource.m in Sources */,
A1BF65D5189A65C6004DD08C /* NSMutableArray+STKAudioPlayer.m in Sources */,

124
StreamingKit/StreamingKit/AEFloatConverter.h Normal file
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@ -0,0 +1,124 @@
//
// AEFloatConverter.h
// The Amazing Audio Engine
//
// Created by Michael Tyson on 25/10/2012.
//
// This software is provided 'as-is', without any express or implied
// warranty. In no event will the authors be held liable for any damages
// arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it
// freely, subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented; you must not
// claim that you wrote the original software. If you use this software
// in a product, an acknowledgment in the product documentation would be
// appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such, and must not be
// misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
#ifdef __cplusplus
extern "C" {
#endif
#import <Foundation/Foundation.h>
#import <AudioToolbox/AudioToolbox.h>
/*!
* Universal converter to float format
*
* Use this class to easily convert arbitrary audio formats to floating point
* for use with utilities like the Accelerate framework.
*/
@interface AEFloatConverter : NSObject
/*!
* Initialize
*
* @param sourceFormat The audio format to use
*/
- (id)initWithSourceFormat:(AudioStreamBasicDescription)sourceFormat;
/*!
* Convert audio to floating-point
*
* This C function, safe to use in a Core Audio realtime thread context, will take
* an audio buffer list of audio in the format you provided at initialisation, and
* convert it into a noninterleaved float array.
*
* @param converter Pointer to the converter object.
* @param sourceBuffer An audio buffer list containing the source audio.
* @param targetBuffers An array of floating-point arrays to store the converted float audio into.
* Note that you must provide the correct number of arrays, to match the number of channels.
* @param frames The number of frames to convert.
* @return YES on success; NO on failure
*/
BOOL AEFloatConverterToFloat(AEFloatConverter* converter, AudioBufferList *sourceBuffer, float * const * targetBuffers, UInt32 frames);
/*!
* Convert audio to floating-point, in a buffer list
*
* This C function, safe to use in a Core Audio realtime thread context, will take
* an audio buffer list of audio in the format you provided at initialisation, and
* convert it into a noninterleaved float format.
*
* @param converter Pointer to the converter object.
* @param sourceBuffer An audio buffer list containing the source audio.
* @param targetBuffer An audio buffer list to store the converted floating-point audio.
* @param frames The number of frames to convert.
* @return YES on success; NO on failure
*/
BOOL AEFloatConverterToFloatBufferList(AEFloatConverter* converter, AudioBufferList *sourceBuffer, AudioBufferList *targetBuffer, UInt32 frames);
/*!
* Convert audio from floating-point
*
* This C function, safe to use in a Core Audio realtime thread context, will take
* an audio buffer list of audio in the format you provided at initialisation, and
* convert it into a float array.
*
* @param converter Pointer to the converter object.
* @param sourceBuffers An array of floating-point arrays containing the floating-point audio to convert.
* Note that you must provide the correct number of arrays, to match the number of channels.
* @param targetBuffer An audio buffer list to store the converted audio into.
* @param frames The number of frames to convert.
* @return YES on success; NO on failure
*/
BOOL AEFloatConverterFromFloat(AEFloatConverter* converter, float * const * sourceBuffers, AudioBufferList *targetBuffer, UInt32 frames);
/*!
* Convert audio from floating-point, in a buffer list
*
* This C function, safe to use in a Core Audio realtime thread context, will take
* an audio buffer list of audio in the format you provided at initialisation, and
* convert it into a float array.
*
* @param converter Pointer to the converter object.
* @param sourceBuffer An audio buffer list containing the source audio.
* @param targetBuffer An audio buffer list to store the converted audio into.
* @param frames The number of frames to convert.
* @return YES on success; NO on failure
*/
BOOL AEFloatConverterFromFloatBufferList(AEFloatConverter* converter, AudioBufferList *sourceBuffer, AudioBufferList *targetBuffer, UInt32 frames);
/*!
* The AudioStreamBasicDescription representing the converted floating-point format
*/
@property (nonatomic, readonly) AudioStreamBasicDescription floatingPointAudioDescription;
/*!
* The source audio format set at initialization
*/
@property (nonatomic, readonly) AudioStreamBasicDescription sourceFormat;
@end
#ifdef __cplusplus
}
#endif

211
StreamingKit/StreamingKit/AEFloatConverter.m Normal file
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@ -0,0 +1,211 @@
//
// AEFloatConverter.m
// The Amazing Audio Engine
//
// Created by Michael Tyson on 25/10/2012.
//
// This software is provided 'as-is', without any express or implied
// warranty. In no event will the authors be held liable for any damages
// arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it
// freely, subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented; you must not
// claim that you wrote the original software. If you use this software
// in a product, an acknowledgment in the product documentation would be
// appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such, and must not be
// misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
#import "AEFloatConverter.h"
#define checkResult(result,operation) (_checkResult((result),(operation),strrchr(__FILE__, '/')+1,__LINE__))
static inline BOOL _checkResult(OSStatus result, const char *operation, const char* file, int line) {
if ( result != noErr ) {
NSLog(@"%s:%d: %s result %d %08X %4.4s", file, line, operation, (int)result, (int)result, (char*)&result);
return NO;
}
return YES;
}
#define kNoMoreDataErr -2222
struct complexInputDataProc_t {
AudioBufferList *sourceBuffer;
};
@interface AEFloatConverter () {
AudioStreamBasicDescription _sourceAudioDescription;
AudioStreamBasicDescription _floatAudioDescription;
AudioConverterRef _toFloatConverter;
AudioConverterRef _fromFloatConverter;
AudioBufferList *_scratchFloatBufferList;
}
static OSStatus complexInputDataProc(AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDescription,
void *inUserData);
@end
@implementation AEFloatConverter
@synthesize sourceFormat = _sourceAudioDescription;
-(id)initWithSourceFormat:(AudioStreamBasicDescription)sourceFormat {
if ( !(self = [super init]) ) return nil;
_floatAudioDescription.mFormatID = kAudioFormatLinearPCM;
_floatAudioDescription.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved;
_floatAudioDescription.mChannelsPerFrame = sourceFormat.mChannelsPerFrame;
_floatAudioDescription.mBytesPerPacket = sizeof(float);
_floatAudioDescription.mFramesPerPacket = 1;
_floatAudioDescription.mBytesPerFrame = sizeof(float);
_floatAudioDescription.mBitsPerChannel = 8 * sizeof(float);
_floatAudioDescription.mSampleRate = sourceFormat.mSampleRate;
_sourceAudioDescription = sourceFormat;
if ( memcmp(&sourceFormat, &_floatAudioDescription, sizeof(AudioStreamBasicDescription)) != 0 ) {
checkResult(AudioConverterNew(&sourceFormat, &_floatAudioDescription, &_toFloatConverter), "AudioConverterNew");
checkResult(AudioConverterNew(&_floatAudioDescription, &sourceFormat, &_fromFloatConverter), "AudioConverterNew");
_scratchFloatBufferList = (AudioBufferList*)malloc(sizeof(AudioBufferList) + (_floatAudioDescription.mChannelsPerFrame-1)*sizeof(AudioBuffer));
_scratchFloatBufferList->mNumberBuffers = _floatAudioDescription.mChannelsPerFrame;
for ( int i=0; i<_scratchFloatBufferList->mNumberBuffers; i++ ) {
_scratchFloatBufferList->mBuffers[i].mNumberChannels = 1;
}
}
return self;
}
-(void)dealloc {
if ( _toFloatConverter ) AudioConverterDispose(_toFloatConverter);
if ( _fromFloatConverter ) AudioConverterDispose(_fromFloatConverter);
if ( _scratchFloatBufferList ) free(_scratchFloatBufferList);
// [super dealloc];
}
BOOL AEFloatConverterToFloat(AEFloatConverter* THIS, AudioBufferList *sourceBuffer, float * const * targetBuffers, UInt32 frames) {
if ( frames == 0 ) return YES;
if ( THIS->_toFloatConverter ) {
UInt32 priorDataByteSize = sourceBuffer->mBuffers[0].mDataByteSize;
for ( int i=0; i<sourceBuffer->mNumberBuffers; i++ ) {
sourceBuffer->mBuffers[i].mDataByteSize = frames * THIS->_sourceAudioDescription.mBytesPerFrame;
}
for ( int i=0; i<THIS->_scratchFloatBufferList->mNumberBuffers; i++ ) {
THIS->_scratchFloatBufferList->mBuffers[i].mData = targetBuffers[i];
THIS->_scratchFloatBufferList->mBuffers[i].mDataByteSize = frames * sizeof(float);
}
OSStatus result = AudioConverterFillComplexBuffer(THIS->_toFloatConverter,
complexInputDataProc,
&(struct complexInputDataProc_t) { .sourceBuffer = sourceBuffer },
&frames,
THIS->_scratchFloatBufferList,
NULL);
for ( int i=0; i<sourceBuffer->mNumberBuffers; i++ ) {
sourceBuffer->mBuffers[i].mDataByteSize = priorDataByteSize;
}
if ( !checkResult(result, "AudioConverterConvertComplexBuffer") ) {
return NO;
}
} else {
for ( int i=0; i<sourceBuffer->mNumberBuffers; i++ ) {
memcpy(targetBuffers[i], sourceBuffer->mBuffers[i].mData, frames * sizeof(float));
}
}
return YES;
}
BOOL AEFloatConverterToFloatBufferList(AEFloatConverter* converter, AudioBufferList *sourceBuffer, AudioBufferList *targetBuffer, UInt32 frames) {
assert(targetBuffer->mNumberBuffers == converter->_floatAudioDescription.mChannelsPerFrame);
float *targetBuffers[targetBuffer->mNumberBuffers];
for ( int i=0; i<targetBuffer->mNumberBuffers; i++ ) {
targetBuffers[i] = (float*)targetBuffer->mBuffers[i].mData;
}
return AEFloatConverterToFloat(converter, sourceBuffer, targetBuffers, frames);
}
BOOL AEFloatConverterFromFloat(AEFloatConverter* THIS, float * const * sourceBuffers, AudioBufferList *targetBuffer, UInt32 frames) {
if ( frames == 0 ) return YES;
if ( THIS->_fromFloatConverter ) {
for ( int i=0; i<THIS->_scratchFloatBufferList->mNumberBuffers; i++ ) {
THIS->_scratchFloatBufferList->mBuffers[i].mData = sourceBuffers[i];
THIS->_scratchFloatBufferList->mBuffers[i].mDataByteSize = frames * sizeof(float);
}
UInt32 priorDataByteSize = targetBuffer->mBuffers[0].mDataByteSize;
for ( int i=0; i<targetBuffer->mNumberBuffers; i++ ) {
targetBuffer->mBuffers[i].mDataByteSize = frames * THIS->_sourceAudioDescription.mBytesPerFrame;
}
OSStatus result = AudioConverterFillComplexBuffer(THIS->_fromFloatConverter,
complexInputDataProc,
&(struct complexInputDataProc_t) { .sourceBuffer = THIS->_scratchFloatBufferList },
&frames,
targetBuffer,
NULL);
for ( int i=0; i<targetBuffer->mNumberBuffers; i++ ) {
targetBuffer->mBuffers[i].mDataByteSize = priorDataByteSize;
}
if ( !checkResult(result, "AudioConverterConvertComplexBuffer") ) {
return NO;
}
} else {
for ( int i=0; i<targetBuffer->mNumberBuffers; i++ ) {
memcpy(targetBuffer->mBuffers[i].mData, sourceBuffers[i], frames * sizeof(float));
}
}
return YES;
}
BOOL AEFloatConverterFromFloatBufferList(AEFloatConverter* converter, AudioBufferList *sourceBuffer, AudioBufferList *targetBuffer, UInt32 frames) {
assert(sourceBuffer->mNumberBuffers == converter->_floatAudioDescription.mChannelsPerFrame);
float *sourceBuffers[sourceBuffer->mNumberBuffers];
for ( int i=0; i<sourceBuffer->mNumberBuffers; i++ ) {
sourceBuffers[i] = (float*)sourceBuffer->mBuffers[i].mData;
}
return AEFloatConverterFromFloat(converter, sourceBuffers, targetBuffer, frames);
}
static OSStatus complexInputDataProc(AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDescription,
void *inUserData) {
struct complexInputDataProc_t *arg = (struct complexInputDataProc_t*)inUserData;
if ( !arg->sourceBuffer ) {
return kNoMoreDataErr;
}
memcpy(ioData, arg->sourceBuffer, sizeof(AudioBufferList) + (arg->sourceBuffer->mNumberBuffers-1)*sizeof(AudioBuffer));
arg->sourceBuffer = NULL;
return noErr;
}
-(AudioStreamBasicDescription)floatingPointAudioDescription {
return _floatAudioDescription;
}
@end

2
StreamingKit/StreamingKit/STKAudioPlayer.h
View File

@ -129,6 +129,8 @@ typedef void(^STKFrameFilter)(UInt32 channelsPerFrame, UInt32 bytesPerFrame, UIn
/// Raised when items queued items are cleared (usually because of a call to play, setDataSource or stop)
-(void) audioPlayer:(STKAudioPlayer*)audioPlayer didCancelQueuedItems:(NSArray*)queuedItems;
-(void) plotGraphWithBuffer:(float*)buffer andLength:(UInt32)count;
@end
@interface STKAudioPlayer : NSObject<STKDataSourceDelegate>

24
StreamingKit/StreamingKit/STKAudioPlayer.m
View File

@ -41,6 +41,7 @@
#import "NSMutableArray+STKAudioPlayer.h"
#import "libkern/OSAtomic.h"
#import <float.h>
#import "AEFloatConverter.h"
#ifndef DBL_MAX
#define DBL_MAX 1.7976931348623157e+308
@ -271,6 +272,9 @@ static AudioStreamBasicDescription recordAudioStreamBasicDescription;
volatile BOOL disposeWasRequested;
volatile BOOL seekToTimeWasRequested;
volatile STKAudioPlayerStopReason stopReason;
float **_floatBuffers;
AEFloatConverter *_floatConverter;
}
@property (readwrite) STKAudioPlayerInternalState internalState;
@ -526,7 +530,21 @@ static void AudioFileStreamPacketsProc(void* clientData, UInt32 numberBytes, UIn
upcomingQueue = [[NSMutableArray alloc] init];
bufferingQueue = [[NSMutableArray alloc] init];
//initialie the float converter
// Allocate the float buffers
_floatConverter = [[AEFloatConverter alloc] initWithSourceFormat:canonicalAudioStreamBasicDescription];
size_t sizeToAllocate = sizeof(float*) * canonicalAudioStreamBasicDescription.mChannelsPerFrame;
sizeToAllocate = MAX(8, sizeToAllocate);
_floatBuffers = (float**)malloc( sizeToAllocate );
UInt32 outputBufferSize = 32 * 1024; // 32 KB
for ( int i=0; i< canonicalAudioStreamBasicDescription.mChannelsPerFrame; i++ ) {
_floatBuffers[i] = (float*)malloc(outputBufferSize);
}
[self resetPcmBuffers];
[self createAudioGraph];
[self createPlaybackThread];
@ -3209,6 +3227,12 @@ static OSStatus OutputRenderCallback(void* inRefCon, AudioUnitRenderActionFlags*
{
[self appendFrameFilterWithName:@"STKMeteringFilter" block:^(UInt32 channelsPerFrame, UInt32 bytesPerFrame, UInt32 frameCount, void* frames)
{
AEFloatConverterToFloat(_floatConverter,&(pcmAudioBufferList),_floatBuffers,frameCount);
if ([self.delegate respondsToSelector:@selector(plotGraphWithBuffer:andLength:)]) {
[self.delegate plotGraphWithBuffer:*(_floatBuffers) andLength:frameCount];
}
SInt16* samples16 = (SInt16*)frames;
SInt32* samples32 = (SInt32*)frames;
UInt32 countLeft = 0;