zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Small refactor on ViE to remove redudant conditions and long ifdefs.

Browse Source 
BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andresp@webrtc.org
2014-08-14 16:46:46 +00:00
parent 2e186386d3
commit a84b0a6dab
3 changed files with 11 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -608,9 +608,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) { int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) {
if (!IsDefaultModule()) { if (!IsDefaultModule()) {
// Don't send from default module. // Don't send from default module.
if (SendingMedia()) { return rtp_sender_.TimeToSendPadding(bytes);
return rtp_sender_.TimeToSendPadding(bytes);
}
} else { } else {
CriticalSectionScoped lock(critical_section_module_ptrs_.get()); CriticalSectionScoped lock(critical_section_module_ptrs_.get());
for (size_t i = 0; i < child_modules_.size(); ++i) { for (size_t i = 0; i < child_modules_.size(); ++i) {

2
webrtc/modules/video_coding/main/interface/video_coding.h
View File

@ -238,7 +238,7 @@ public:
// frame rate/dimensions need to be updated for video quality optimization // frame rate/dimensions need to be updated for video quality optimization
// //
// Input: // Input:
// - videoQMSettings : The callback object to register. // - videoQMSettings : The callback object to register.
// //
// Return value : VCM_OK, on success. // Return value : VCM_OK, on success.
// < 0, on error // < 0, on error

29
webrtc/video_engine/vie_encoder.cc
View File

@ -187,8 +187,13 @@ bool ViEEncoder::Init() {
qm_callback_ = new QMVideoSettingsCallback(&vpm_); qm_callback_ = new QMVideoSettingsCallback(&vpm_);
#ifdef VIDEOCODEC_VP8 #ifdef VIDEOCODEC_VP8
VideoCodecType codec_type = webrtc::kVideoCodecVP8;
#else
VideoCodecType codec_type = webrtc::kVideoCodecI420;
#endif
VideoCodec video_codec; VideoCodec video_codec;
if (vcm_.Codec(webrtc::kVideoCodecVP8, &video_codec) != VCM_OK) { if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
return false; return false;
} }
{ {
@ -202,21 +207,6 @@ bool ViEEncoder::Init() {
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) { if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
return false; return false;
} }
#else
VideoCodec video_codec;
if (vcm_.Codec(webrtc::kVideoCodecI420, &video_codec) == VCM_OK) {
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
default_rtp_rtcp_->MaxDataPayloadLength());
default_rtp_rtcp_->RegisterSendPayload(video_codec);
} else {
return false;
}
#endif
if (vcm_.RegisterTransportCallback(this) != 0) { if (vcm_.RegisterTransportCallback(this) != 0) {
return false; return false;
} }
@ -362,11 +352,10 @@ int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
// Set this module as sending right away, let the slave module in the channel // Set this module as sending right away, let the slave module in the channel
// start and stop sending. // start and stop sending.
if (default_rtp_rtcp_->Sending() == false) { if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
if (default_rtp_rtcp_->SetSendingStatus(true) != 0) { return -1;
return -1;
}
} }
bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(), bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
video_codec.startBitrate * 1000, video_codec.startBitrate * 1000,
video_codec.minBitrate * 1000, video_codec.minBitrate * 1000,