forked from chrisballinger/FFmpeg-iOS-Encoder
-
Notifications
You must be signed in to change notification settings - Fork 0
Expand file tree
/
Copy pathAACEncoder.m
More file actions
215 lines (190 loc) · 10.1 KB
/
Copy pathAACEncoder.m
File metadata and controls
215 lines (190 loc) · 10.1 KB
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
//
// AACEncoder.m
// FFmpegEncoder
//
// Created by Christopher Ballinger on 12/18/13.
// Copyright (c) 2013 Christopher Ballinger. All rights reserved.
//
// http://stackoverflow.com/questions/10817036/can-i-use-avcapturesession-to-encode-an-aac-stream-to-memory
#import "AACEncoder.h"
#import <AVFoundation/AVFoundation.h>
#import <AudioToolbox/AudioToolbox.h>
@interface AACEncoder()
@property (nonatomic) AudioConverterRef audioConverter;
@property (nonatomic) uint8_t *aacBuffer;
@property (nonatomic) NSUInteger aacBufferSize;
@property (nonatomic) char *pcmBuffer;
@property (nonatomic) size_t pcmBufferSize;
@end
@implementation AACEncoder
- (void) dealloc {
AudioConverterDispose(_audioConverter);
free(_aacBuffer);
}
- (id) init {
if (self = [super init]) {
_encoderQueue = dispatch_queue_create("AAC Encoder Queue", DISPATCH_QUEUE_SERIAL);
_callbackQueue = dispatch_queue_create("AAC Encoder Callback Queue", DISPATCH_QUEUE_SERIAL);
_audioConverter = NULL;
_pcmBufferSize = 0;
_pcmBuffer = NULL;
_aacBufferSize = 1024;
_aacBuffer = malloc(_aacBufferSize * sizeof(uint8_t));
memset(_aacBuffer, 0, _aacBufferSize);
}
return self;
}
- (void) setupEncoderFromSampleBuffer:(CMSampleBufferRef)sampleBuffer {
AudioStreamBasicDescription inAudioStreamBasicDescription = *CMAudioFormatDescriptionGetStreamBasicDescription((CMAudioFormatDescriptionRef)CMSampleBufferGetFormatDescription(sampleBuffer));
AudioStreamBasicDescription outAudioStreamBasicDescription = {0}; // Always initialize the fields of a new audio stream basic description structure to zero, as shown here: ...
outAudioStreamBasicDescription.mSampleRate = inAudioStreamBasicDescription.mSampleRate; // The number of frames per second of the data in the stream, when the stream is played at normal speed. For compressed formats, this field indicates the number of frames per second of equivalent decompressed data. The mSampleRate field must be nonzero, except when this structure is used in a listing of supported formats (see “kAudioStreamAnyRate”).
outAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC; // kAudioFormatMPEG4AAC_HE does not work. Can't find `AudioClassDescription`. `mFormatFlags` is set to 0.
outAudioStreamBasicDescription.mFormatFlags = kMPEG4Object_AAC_LC; // Format-specific flags to specify details of the format. Set to 0 to indicate no format flags. See “Audio Data Format Identifiers” for the flags that apply to each format.
outAudioStreamBasicDescription.mBytesPerPacket = 0; // The number of bytes in a packet of audio data. To indicate variable packet size, set this field to 0. For a format that uses variable packet size, specify the size of each packet using an AudioStreamPacketDescription structure.
outAudioStreamBasicDescription.mFramesPerPacket = 1024; // The number of frames in a packet of audio data. For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC. For formats with a variable number of frames per packet, such as Ogg Vorbis, set this field to 0.
outAudioStreamBasicDescription.mBytesPerFrame = 0; // The number of bytes from the start of one frame to the start of the next frame in an audio buffer. Set this field to 0 for compressed formats. ...
outAudioStreamBasicDescription.mChannelsPerFrame = 1; // The number of channels in each frame of audio data. This value must be nonzero.
outAudioStreamBasicDescription.mBitsPerChannel = 0; // ... Set this field to 0 for compressed formats.
outAudioStreamBasicDescription.mReserved = 0; // Pads the structure out to force an even 8-byte alignment. Must be set to 0.
AudioClassDescription *description = [self
getAudioClassDescriptionWithType:kAudioFormatMPEG4AAC
fromManufacturer:kAppleSoftwareAudioCodecManufacturer];
OSStatus status = AudioConverterNewSpecific(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, 1, description, &_audioConverter);
if (status != 0) {
NSLog(@"setup converter: %d", (int)status);
}
}
- (AudioClassDescription *)getAudioClassDescriptionWithType:(UInt32)type
fromManufacturer:(UInt32)manufacturer
{
static AudioClassDescription desc;
UInt32 encoderSpecifier = type;
OSStatus st;
UInt32 size;
st = AudioFormatGetPropertyInfo(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size);
if (st) {
NSLog(@"error getting audio format propery info: %d", (int)(st));
return nil;
}
unsigned int count = size / sizeof(AudioClassDescription);
AudioClassDescription descriptions[count];
st = AudioFormatGetProperty(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size,
descriptions);
if (st) {
NSLog(@"error getting audio format propery: %d", (int)(st));
return nil;
}
for (unsigned int i = 0; i < count; i++) {
if ((type == descriptions[i].mSubType) &&
(manufacturer == descriptions[i].mManufacturer)) {
memcpy(&desc, &(descriptions[i]), sizeof(desc));
return &desc;
}
}
return nil;
}
static OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
AACEncoder *encoder = (__bridge AACEncoder *)(inUserData);
UInt32 requestedPackets = *ioNumberDataPackets;
//NSLog(@"Number of packets requested: %d", (unsigned int)requestedPackets);
size_t copiedSamples = [encoder copyPCMSamplesIntoBuffer:ioData];
if (copiedSamples < requestedPackets) {
//NSLog(@"PCM buffer isn't full enough!");
*ioNumberDataPackets = 0;
return -1;
}
*ioNumberDataPackets = 1;
//NSLog(@"Copied %zu samples into ioData", copiedSamples);
return noErr;
}
- (size_t) copyPCMSamplesIntoBuffer:(AudioBufferList*)ioData {
size_t originalBufferSize = _pcmBufferSize;
if (!originalBufferSize) {
return 0;
}
ioData->mBuffers[0].mData = _pcmBuffer;
ioData->mBuffers[0].mDataByteSize = _pcmBufferSize;
_pcmBuffer = NULL;
_pcmBufferSize = 0;
return originalBufferSize;
}
- (void) encodeSampleBuffer:(CMSampleBufferRef)sampleBuffer completionBlock:(void (^)(NSData * encodedData, NSError* error))completionBlock {
CFRetain(sampleBuffer);
dispatch_async(_encoderQueue, ^{
if (!_audioConverter) {
[self setupEncoderFromSampleBuffer:sampleBuffer];
}
CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
CFRetain(blockBuffer);
OSStatus status = CMBlockBufferGetDataPointer(blockBuffer, 0, NULL, &_pcmBufferSize, &_pcmBuffer);
NSError *error = nil;
if (status != kCMBlockBufferNoErr) {
error = [NSError errorWithDomain:NSOSStatusErrorDomain code:status userInfo:nil];
}
//NSLog(@"PCM Buffer Size: %zu", _pcmBufferSize);
memset(_aacBuffer, 0, _aacBufferSize);
AudioBufferList outAudioBufferList = {0};
outAudioBufferList.mNumberBuffers = 1;
outAudioBufferList.mBuffers[0].mNumberChannels = 1;
outAudioBufferList.mBuffers[0].mDataByteSize = _aacBufferSize;
outAudioBufferList.mBuffers[0].mData = _aacBuffer;
AudioStreamPacketDescription *outPacketDescription = NULL;
UInt32 ioOutputDataPacketSize = 1;
status = AudioConverterFillComplexBuffer(_audioConverter, inInputDataProc, (__bridge void *)(self), &ioOutputDataPacketSize, &outAudioBufferList, outPacketDescription);
//NSLog(@"ioOutputDataPacketSize: %d", (unsigned int)ioOutputDataPacketSize);
NSData *data = nil;
if (status == 0) {
NSData *rawAAC = [NSData dataWithBytes:outAudioBufferList.mBuffers[0].mData length:outAudioBufferList.mBuffers[0].mDataByteSize];
NSData *adtsHeader = [self adtsDataForPacketLength:rawAAC.length];
NSMutableData *fullData = [NSMutableData dataWithData:adtsHeader];
[fullData appendData:rawAAC];
data = fullData;
} else {
error = [NSError errorWithDomain:NSOSStatusErrorDomain code:status userInfo:nil];
}
if (completionBlock) {
dispatch_async(_callbackQueue, ^{
completionBlock(data, error);
});
}
CFRelease(sampleBuffer);
CFRelease(blockBuffer);
});
}
/**
* Add ADTS header at the beginning of each and every AAC packet.
* This is needed as MediaCodec encoder generates a packet of raw
* AAC data.
*
* Note the packetLen must count in the ADTS header itself.
* See: http://wiki.multimedia.cx/index.php?title=ADTS
* Also: http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio#Channel_Configurations
**/
- (NSData*) adtsDataForPacketLength:(NSUInteger)packetLength {
int adtsLength = 7;
char *packet = malloc(sizeof(char) * adtsLength);
// Variables Recycled by addADTStoPacket
int profile = 2; //AAC LC
//39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 4; //44.1KHz
int chanCfg = 1; //MPEG-4 Audio Channel Configuration. 1 Channel front-center
NSUInteger fullLength = adtsLength + packetLength;
// fill in ADTS data
packet[0] = (char)0xFF; // 11111111 = syncword
packet[1] = (char)0xF9; // 1111 1 00 1 = syncword MPEG-2 Layer CRC
packet[2] = (char)(((profile-1)<<6) + (freqIdx<<2) +(chanCfg>>2));
packet[3] = (char)(((chanCfg&3)<<6) + (fullLength>>11));
packet[4] = (char)((fullLength&0x7FF) >> 3);
packet[5] = (char)(((fullLength&7)<<5) + 0x1F);
packet[6] = (char)0xFC;
NSData *data = [NSData dataWithBytesNoCopy:packet length:adtsLength freeWhenDone:YES];
return data;
}
@end