WebRtc_UWord8
TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecID_B)
{
AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
WebRtc_UWord8 noCodec = tmpACM->NumberOfCodecs();
CodecInst codecInst;
printf("List of Supported Codecs\n");
printf("========================\n");
for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
{
tmpACM->Codec(codecCntr, codecInst);
printf("%d- %s\n", codecCntr, codecInst.plname);
}
printf("\nChoose a send codec for side A [0]: ");
char myStr[15] = "";
fgets(myStr, 10, stdin);
*codecID_A = (WebRtc_UWord8)atoi(myStr);
printf("\nChoose a send codec for side B [0]: ");
fgets(myStr, 10, stdin);
*codecID_B = (WebRtc_UWord8)atoi(myStr);
AudioCodingModule::Destroy(tmpACM);
printf("\n");
return 0;
}
static pj_status_t webrtc_enum_codecs(pjmedia_codec_factory *factory,
unsigned *count, pjmedia_codec_info codecs[]) {
unsigned max;
unsigned i;
int numCodecs = 1;
PJ_UNUSED_ARG(factory);
PJ_ASSERT_RETURN(codecs && *count > 0, PJ_EINVAL);
max = *count;
// TODO : we could use AudioCodingModule::RegisterReceiveCodec / UnRegister... in order to change PT
AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
struct CodecInst sendCodecTmp;
numCodecs = acmTmp->NumberOfCodecs();
PJ_LOG(4, (THIS_FILE, "List of supported codec."));
for (i = 0, *count = 0; i < numCodecs && *count < max; ++i) {
acmTmp->Codec(i, sendCodecTmp);
pj_str_t codec_name = pj_str((char*) sendCodecTmp.plname);
// Exclude useless codecs
if ((pj_stricmp2(&codec_name, "telephone-event") != 0)
&& (pj_stricmp2(&codec_name, "cn") != 0)
/* Exclude PCMU/PCMA cause already in pjsip and wertc force that to be built in*/
&& (pj_stricmp2(&codec_name, "pcmu") != 0)
&& (pj_stricmp2(&codec_name, "pcma") != 0)
) {
PJ_LOG(
4,
(THIS_FILE, "%d %s %d %d %d %d", i, sendCodecTmp.plname, sendCodecTmp.pltype, sendCodecTmp.plfreq, sendCodecTmp.pacsize, sendCodecTmp.rate));
pj_bzero(&codecs[*count], sizeof(pjmedia_codec_info));
pj_strdup2(webrtc_factory.pool, &codecs[*count].encoding_name,
sendCodecTmp.plname);
codecs[*count].pt = sendCodecTmp.pltype;
codecs[*count].type = PJMEDIA_TYPE_AUDIO;
codecs[*count].clock_rate = sendCodecTmp.plfreq;
codecs[*count].channel_cnt = sendCodecTmp.channels;
++*count;
}
}
AudioCodingModule::Destroy(acmTmp);
return PJ_SUCCESS;
}
void EncodeToFileTest::Perform(int fileType, int codeId, int* codePars, int testMode)
{
AudioCodingModule *acm = AudioCodingModule::Create(0);
RTPFile rtpFile;
char fileName[] = "outFile.rtp";
rtpFile.Open(fileName, "wb+");
rtpFile.WriteHeader();
//for auto_test and logging
_sender.testMode = testMode;
_sender.codeId = codeId;
_sender.Setup(acm, &rtpFile);
struct CodecInst sendCodecInst;
if(acm->SendCodec(sendCodecInst) >= 0)
{
_sender.Run();
}
_sender.Teardown();
rtpFile.Close();
AudioCodingModule::Destroy(acm);
}