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Remove SetCaptureDelay from the RTP module.

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This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2015-02-17 16:36:08 +00:00
parent 7663684258
commit 0abc6011b9
7 changed files with 0 additions and 47 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

7
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
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@ -610,13 +610,6 @@ class RtpRtcp : public Module {
*
***************************************************************************/
/*
* Set the estimated camera delay in MS
*
* return -1 on failure else 0
*/
virtual int32_t SetCameraDelay(int32_t delayMS) = 0;
/*
* Set the target send bitrate
*/

2
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -231,8 +231,6 @@ class MockRtpRtcp : public RtpRtcp {
int32_t(bool& enable, uint8_t& ID));
MOCK_METHOD1(SetAudioLevel,
int32_t(const uint8_t level_dBov));
MOCK_METHOD1(SetCameraDelay,
int32_t(const int32_t delayMS));
MOCK_METHOD1(SetTargetSendBitrate,
void(const std::vector<uint32_t>& stream_bitrates));
MOCK_METHOD3(SetGenericFECStatus,

15
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -108,9 +108,6 @@ RTCPSender::RTCPSender(int32_t id,
internal_report_blocks_(),
external_report_blocks_(),
_csrcCNAMEs(),
_cameraDelayMS(0),
_lastSendReport(),
_lastRTCPTime(),
@ -302,18 +299,6 @@ void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
_remoteSSRC = ssrc;
}
int32_t RTCPSender::SetCameraDelay(int32_t delayMS) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(delayMS > 1000 || delayMS < -1000)
{
LOG(LS_WARNING) << "Delay can't be larger than 1 second: "
<< delayMS << " ms";
return -1;
}
_cameraDelayMS = delayMS;
return 0;
}
int32_t RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
if (!cName)
return -1;

4
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -93,8 +93,6 @@ public:
void SetRemoteSSRC(uint32_t ssrc);
int32_t SetCameraDelay(int32_t delayMS);
int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]);
int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]);
@ -303,8 +301,6 @@ private:
std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs
GUARDED_BY(_criticalSectionRTCPSender);
int32_t _cameraDelayMS GUARDED_BY(_criticalSectionRTCPSender);
// Sent
uint32_t _lastSendReport[RTCP_NUMBER_OF_SR] GUARDED_BY(
_criticalSectionRTCPSender); // allow packet loss and RTT above 1 sec

16
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -959,22 +959,6 @@ int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id);
}
int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) {
if (IsDefaultModule()) {
CriticalSectionScoped lock(critical_section_module_ptrs_.get());
std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
while (it != child_modules_.end()) {
RtpRtcp* module = *it;
if (module) {
module->SetCameraDelay(delay_ms);
}
it++;
}
return 0;
}
return rtcp_sender_.SetCameraDelay(delay_ms);
}
int32_t ModuleRtpRtcpImpl::SetGenericFECStatus(
const bool enable,
const uint8_t payload_type_red,

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -292,8 +292,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Send a request for a keyframe.
virtual int32_t RequestKeyFrame() OVERRIDE;
virtual int32_t SetCameraDelay(int32_t delay_ms) OVERRIDE;
virtual void SetTargetSendBitrate(
const std::vector<uint32_t>& stream_bitrates) OVERRIDE;

1
webrtc/video_engine/vie_encoder.cc
View File

@ -618,7 +618,6 @@ void ViEEncoder::DeliverFrame(int id,
}
void ViEEncoder::DelayChanged(int id, int frame_delay) {
default_rtp_rtcp_->SetCameraDelay(frame_delay);
}
int ViEEncoder::GetPreferedFrameSettings(int* width,