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Break out RTCPSender dependency on ModuleRtpRtcpImpl.

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BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-09-09 16:02:19 +00:00
parent 26b0d77baf
commit 59f20bb735
7 changed files with 158 additions and 123 deletions
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27
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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@ -232,7 +232,9 @@ int32_t RTPSender::DeRegisterSendPayload(
int8_t RTPSender::SendPayloadType() const { return payload_type_; }
int RTPSender::SendPayloadFrequency() const { return audio_->AudioFrequency(); }
int RTPSender::SendPayloadFrequency() const {
return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
}
int32_t RTPSender::SetMaxPayloadLength(
const uint16_t max_payload_length,
@ -1168,28 +1170,9 @@ bool RTPSender::UpdateAbsoluteSendTime(
return true;
}
void RTPSender::SetSendingStatus(const bool enabled) {
void RTPSender::SetSendingStatus(bool enabled) {
if (enabled) {
uint32_t frequency_hz;
if (audio_configured_) {
uint32_t frequency = audio_->AudioFrequency();
// sanity
switch (frequency) {
case 8000:
case 12000:
case 16000:
case 24000:
case 32000:
break;
default:
assert(false);
return;
}
frequency_hz = frequency;
} else {
frequency_hz = kVideoPayloadTypeFrequency;
}
uint32_t frequency_hz = SendPayloadFrequency();
uint32_t RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_, frequency_hz);
// Will be ignored if it's already configured via API.