[Constructor]
WebRtcAudioSendStream(
uint32_t ssrc,
const std::string &mid,
const std::string &c_name,
const std::string track_id,
const adsl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>& send_codec_spec,
int max_send_bitrate_bps,
const adsl::optional<std::string>& audio_network_adaptor_config,
webrtc::Call call,
webrtc:: Transport send_transport,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory> & encoder_factory,
const adsl::optional<webrtc::AudioCodecPairId> codec_pairid) :
call(call),
config_(send_transport),
send_side_bwe_withoverhead(webrtc::field_trial::IsEnabled("WebRtc-SendSideBwe-WithOverHead")),
max_send_bitratebps(max_send_bitrate_bps),
rtpparameters (CreateParametersWithOneEncoding())
[CONFIG SETUP]
[config_(sendtransport)] [use Transport structure]
config.rtp.ssrc=ssrc;
config.rtp.mid =mid;
config.rtp.c_name=cname;
config.rtp.extensions=extensions;
config_.audio_network_adaptor_config= audio_network_adaptorconfig
config.encoder_factory= encoderfactory;
config.codec_pair_id=codec_pairid;
config.track_id=track_id
[rtp_paramters (CreateParametersWithOneEncoding)]
rtpparamters.encodings[0].ssrc=ssrc;
rtpparamters.rtcp.name=c_name;
rtpparamters.header_extensions=extensions;
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