1、VoiceEngine Codec数据结构
WebRTC中,用一个结构体struct CodecInst表示特定的音频编解码器对象:
- struct CodecInst
- {
- int pltype; //payload type负载类型
- char plname[32]; //payload name负载名称,32个字符表示
- int plfreq; //payload frequence负载频率
- int pacsize; //packet size包大小
- int channels; //声道
- int rate; //速率或自适应
- };
参数详细说明:
1、 pltype范围在1~126之间才是有效值;
pltype的值是否有效可以通过调用下面ValidPayloadType(int payload_type)方法来判断,在...\src\modules\audio_coding\main\source\acm_codec_database.cc定义
- // Checks if the payload type is in the valid range.
- bool ACMCodecDB::ValidPayloadType(int payload_type) {
- if ((payload_type < 0) || (payload_type > 127)) {
- return false;
- }
- return true;
- }
2、 plname是编解码器的名称,可能的值在CreateCodecInstance已定义,如WebRTC默认的"ISAC"
VoiceEngine支持多个音频编解码器,具体支持的编解码器在CreateCodecInstance(const CodecInst* codec_inst)定义,比如ISAC\PCMU\PCMA\ILBC\AMR等等,在...\src\modules\audio_coding\main\source\acm_codec_database.cc定义
- ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
- // All we have support for right now.
- if (!STR_CASE_CMP(codec_inst->plname, "ISAC")) {
- #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- return new ACMISAC(kISAC);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "PCMU")) {
- return new ACMPCMU(kPCMU);
- } else if (!STR_CASE_CMP(codec_inst->plname, "PCMA")) {
- return new ACMPCMA(kPCMA);
- } else if (!STR_CASE_CMP(codec_inst->plname, "ILBC")) {
- #ifdef WEBRTC_CODEC_ILBC
- return new ACMILBC(kILBC);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "AMR")) {
- #ifdef WEBRTC_CODEC_AMR
- return new ACMAMR(kGSMAMR);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "AMR-WB")) {
- #ifdef WEBRTC_CODEC_AMRWB
- return new ACMAMRwb(kGSMAMRWB);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G722")) {
- #ifdef WEBRTC_CODEC_G722
- return new ACMG722(kG722);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G7221")) {
- switch (codec_inst->plfreq) {
- case 16000: {
- #ifdef WEBRTC_CODEC_G722_1
- int codec_id;
- switch (codec_inst->rate) {
- case 16000 : {
- codec_id = kG722_1_16;
- break;
- }
- case 24000 : {
- codec_id = kG722_1_24;
- break;
- }
- case 32000 : {
- codec_id = kG722_1_32;
- break;
- }
- default: {
- return NULL;
- }
- return new ACMG722_1(codec_id);
- }
- #endif
- }
- case 32000: {
- #ifdef WEBRTC_CODEC_G722_1C
- int codec_id;
- switch (codec_inst->rate) {
- case 24000 : {
- codec_id = kG722_1C_24;
- break;
- }
- case 32000 : {
- codec_id = kG722_1C_32;
- break;
- }
- case 48000 : {
- codec_id = kG722_1C_48;
- break;
- }
- default: {
- return NULL;
- }
- return new ACMG722_1C(codec_id);
- }
- #endif
- }
- }
- } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
- // For CN we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kCNNB;
- break;
- }
- case 16000: {
- codec_id = kCNWB;
- break;
- }
- case 32000: {
- codec_id = kCNSWB;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMCNG(codec_id);
- } else if (!STR_CASE_CMP(codec_inst->plname, "G729")) {
- #ifdef WEBRTC_CODEC_G729
- return new ACMG729(kG729);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G7291")) {
- #ifdef WEBRTC_CODEC_G729_1
- return new ACMG729_1(kG729_1);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
- #ifdef WEBRTC_CODEC_SPEEX
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kSPEEX8;
- break;
- }
- case 16000: {
- codec_id = kSPEEX16;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMSPEEX(codec_id);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
- // For CN we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kCNNB;
- break;
- }
- case 16000: {
- codec_id = kCNWB;
- break;
- }
- case 32000: {
- codec_id = kCNSWB;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMCNG(codec_id);
- } else if (!STR_CASE_CMP(codec_inst->plname, "L16")) {
- #ifdef WEBRTC_CODEC_PCM16
- // For L16 we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kPCM16B;
- break;
- }
- case 16000: {
- codec_id =kPCM16Bwb;
- break;
- }
- case 32000: {
- codec_id = kPCM16Bswb32kHz;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMPCM16B(codec_id);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "telephone-event")) {
- #ifdef WEBRTC_CODEC_AVT
- return new ACMDTMFPlayout(kAVT);
- #endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "red")) {
- #ifdef WEBRTC_CODEC_RED
- return new ACMRED(kRED);
- #endif
- }
- return NULL;
- }
3、 plfreq一般取如下值(在common_types.h定义);
- //负载频率值
- enum PayloadFrequencies
- {
- kFreq8000Hz = 8000,
- kFreq16000Hz = 16000,
- kFreq32000Hz = 32000
- };
4、 pacsize取值是与plfreq有关系的,单位为kbps,下面是计算公式
计算公式如下:
如果:plfreq = 16000(单位为hz)
如果我需要30ms(毫秒)的packet size
那么pacsize = (plfreq *30) /1000 = 480kbps;
也即是:要得到k ms的packet size,则可计算出
pacsize =( plfreq * k) / 1000
而如果plfreq = 32000;20ms的packet size,则pacsize = 640;
5、 channels取值
channels = 1 表示单声道
channels = 2 表示立体声道
注意:channels = -1时,表示此时只支持单声道模式
6、 rate取值,单位是bps
一般取rate = 16000,32000,48000这些16000整数倍的值,即16kbps,32kbps,48kpbs
注意:当rate = -1时,表示此时启动自适应信道速率
2、查看VoiceEngine支持的所有Codec信息示例代码
- //列出(获得)引擎支持的所有编解码器信息
- //支持平台:Windows, Mac OS X, Linux
- #include "voe_base.h"
- #include "voe_codec.h"
- VoiceEngine* ve = VoiceEngine::Create();
- VoECodec* codec = VoECodec::GetInterface(ve);
- for (int = 0; i < codec->NumOfCodecs(); i++)
- {
- CodecInst cinst;
- codec->GetCodec(i, cinst);
- DISPLAY_CODEC_INFO(i, cinst);
- }
- // 释放sub-API
- codec->Release();
- //删除引擎
- VoiceEngine::Delete(ve);
3、初始化VoiceEngine Codec示例代码
- //初始化VoiceEngine Codec示例代码
- //支持平台:Windows, Mac OS X, Linux
- #include "voe_codec.h"
- CodecInst cinst;
- //初始化iSAC编解码器参数
- strcpy(cinst.plname, "ISAC");
- cinst.plfreq = 16000; // iSAC宽带模式取样频率
- cinst.pltype = 103;
- cinst.pacsize = 480; //使用30ms packet size,480kbps
- cinst.channels = 1; // 单声道
- cinst.rate = -1; // 信道自适应模式
- //初始化完成
- //在ID为0的channel激活已初始化的iSAC
- codec->SetSendCodec(0, cinst);
WebRTC音视频引擎研究(2)--VoiceEngine音频编解码器数据结构以及参数设置,布布扣,bubuko.com