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Ant Media Server supports RTMP, RTSP, MP4, HLS, WebRTC, Adaptive Streaming, etc.
package io.antmedia.muxer.parser;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
public class AACConfigParser extends Parser {
protected static Logger logger = LoggerFactory.getLogger(AACConfigParser.class);
/*
0: 96000 Hz
1: 88200 Hz
2: 64000 Hz
3: 48000 Hz
4: 44100 Hz
5: 32000 Hz
6: 24000 Hz
7: 22050 Hz
8: 16000 Hz
9: 12000 Hz
10: 11025 Hz
11: 8000 Hz
12: 7350 Hz
13: Reserved
14: Reserved
15: frequency is written explictly
*/
protected static final int SAMPLE_RATE_96000 = 0;
protected static final int SAMPLE_RATE_88200 = 1;
protected static final int SAMPLE_RATE_64000 = 2;
protected static final int SAMPLE_RATE_48000 = 3;
protected static final int SAMPLE_RATE_44100 = 4;
protected static final int SAMPLE_RATE_32000 = 5;
protected static final int SAMPLE_RATE_24000 = 6;
protected static final int SAMPLE_RATE_22050 = 7;
protected static final int SAMPLE_RATE_16000 = 8;
protected static final int SAMPLE_RATE_12000 = 9;
protected static final int SAMPLE_RATE_11025 = 10;
protected static final int SAMPLE_RATE_8000 = 11;
protected static final int SAMPLE_RATE_7350 = 12;
private int sampleRate;
/*
0: Defined in AOT Specifc Config
1: 1 channel: front-center
2: 2 channels: front-left, front-right
3: 3 channels: front-center, front-left, front-right
4: 4 channels: front-center, front-left, front-right, back-center
5: 5 channels: front-center, front-left, front-right, back-left, back-right
6: 6 channels: front-center, front-left, front-right, back-left, back-right, LFE-channel
7: 8 channels: front-center, front-left, front-right, side-left, side-right, back-left, back-right, LFE-channel
8-15: Reserved
*/
private int channelCount;
/*
0: Null
1: AAC Main
2: AAC LC (Low Complexity)
3: AAC SSR (Scalable Sample Rate)
4: AAC LTP (Long Term Prediction)
5: SBR (Spectral Band Replication)
6: AAC Scalable
7: TwinVQ
8: CELP (Code Excited Linear Prediction)
9: HXVC (Harmonic Vector eXcitation Coding)
10: Reserved
11: Reserved
12: TTSI (Text-To-Speech Interface)
13: Main Synthesis
14: Wavetable Synthesis
15: General MIDI
16: Algorithmic Synthesis and Audio Effects
17: ER (Error Resilient) AAC LC
18: Reserved
19: ER AAC LTP
20: ER AAC Scalable
21: ER TwinVQ
22: ER BSAC (Bit-Sliced Arithmetic Coding)
23: ER AAC LD (Low Delay)
24: ER CELP
25: ER HVXC
26: ER HILN (Harmonic and Individual Lines plus Noise)
27: ER Parametric
28: SSC (SinuSoidal Coding)
29: PS (Parametric Stereo)
30: MPEG Surround
31: (Escape value)
32: Layer-1
33: Layer-2
34: Layer-3
35: DST (Direct Stream Transfer)
36: ALS (Audio Lossless)
37: SLS (Scalable LosslesS)
38: SLS non-core
39: ER AAC ELD (Enhanced Low Delay)
40: SMR (Symbolic Music Representation) Simple
41: SMR Main
42: USAC (Unified Speech and Audio Coding) (no SBR)
43: SAOC (Spatial Audio Object Coding)
44: LD MPEG Surround
45: USAC
*/
public enum AudioObjectTypes {
NULL(0),
AAC_MAIN(1),
AAC_LC(2),
AAC_SSR(3),
AAC_LTP(4),
SBR(5),
AAC_SCALABLE(6);
public final int value;
AudioObjectTypes(int value) {
this.value = value;
}
}
private AudioObjectTypes objectType;
private int frameSize;
public AACConfigParser(byte[] data, int offset)
{
super(data, offset);
}
protected void parse() {
/**
5 bits: object type
if (object type == 31)
6 bits + 32: object type
4 bits: frequency index
if (frequency index == 15)
24 bits: frequency
4 bits: channel configuration
var bits: AOT Specific Config
*/
int objectTypeIndex = readBits(5);
if (objectTypeIndex == AudioObjectTypes.NULL.value) {
objectType = AudioObjectTypes.NULL;
}
else if (objectTypeIndex == AudioObjectTypes.AAC_MAIN.value) {
objectType = AudioObjectTypes.AAC_MAIN;
}
else if (objectTypeIndex == AudioObjectTypes.AAC_LC.value) {
objectType = AudioObjectTypes.AAC_LC;
}
else if (objectTypeIndex == AudioObjectTypes.AAC_SSR.value) {
objectType = AudioObjectTypes.AAC_SSR;
}
else if (objectTypeIndex == AudioObjectTypes.AAC_LTP.value) {
objectType = AudioObjectTypes.AAC_LTP;
}
else if (objectTypeIndex == AudioObjectTypes.SBR.value) {
objectType = AudioObjectTypes.SBR;
}
else if (objectTypeIndex == AudioObjectTypes.AAC_SCALABLE.value) {
objectType = AudioObjectTypes.AAC_SCALABLE;
}
else {
logger.error("Cannot determine the AAC object type:{} ", objectTypeIndex);
errorOccured = true;
return;
}
if (objectType == AudioObjectTypes.NULL) {
errorOccured = true;
logger.error("Cannot determine the AAC object type it's null ");
return;
}
logger.info("AAC object type:{} ", objectType);
int sampleRateIndex = readBits(4);
if (sampleRateIndex == SAMPLE_RATE_96000) {
sampleRate = 96000;
}
else if (sampleRateIndex == SAMPLE_RATE_88200) {
sampleRate = 88200;
}
else if (sampleRateIndex == SAMPLE_RATE_64000) {
sampleRate = 64000;
}
else if (sampleRateIndex == SAMPLE_RATE_48000) {
sampleRate = 48000;
}
else if (sampleRateIndex == SAMPLE_RATE_44100) {
sampleRate = 44100;
}
else if (sampleRateIndex == SAMPLE_RATE_32000) {
sampleRate = 32000;
}
else if (sampleRateIndex == SAMPLE_RATE_24000) {
sampleRate = 24000;
}
else if (sampleRateIndex == SAMPLE_RATE_22050) {
sampleRate = 22050;
}
else if (sampleRateIndex == SAMPLE_RATE_16000) {
sampleRate = 16000;
}
else if (sampleRateIndex == SAMPLE_RATE_12000) {
sampleRate = 12000;
}
else if (sampleRateIndex == SAMPLE_RATE_11025) {
sampleRate = 11025;
}
else if (sampleRateIndex == SAMPLE_RATE_8000) {
sampleRate = 8000;
}
else if (sampleRateIndex == SAMPLE_RATE_7350) {
sampleRate = 7350;
}
else {
logger.error("Cannot determine the AAC Sample Rate:{} ", sampleRateIndex);
errorOccured = true;
return;
}
logger.info("AAC Sample rate:{} ", sampleRate);
channelCount = readBits(4);
if (channelCount == 0 || channelCount > 7) {
logger.error("Cannot determine the channel count: {}", channelCount);
errorOccured = true;
return;
}
if (channelCount == 7) {
channelCount = 8;
}
frameSize = readBit() == 0x00 ? 1024 : 960;
}
public AudioObjectTypes getObjectType() {
return objectType;
}
public int getChannelCount() {
return channelCount;
}
public int getSampleRate() {
return sampleRate;
}
public int getFrameSize() {
return frameSize;
}
}