From a84b0a6dabdf5c0c6f120bd72ad15653a0d3ddcf Mon Sep 17 00:00:00 2001 From: "andresp@webrtc.org" Date: Thu, 14 Aug 2014 16:46:46 +0000 Subject: [PATCH] Small refactor on ViE to remove redudant conditions and long ifdefs. 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 --- .../modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 4 +-- .../main/interface/video_coding.h | 2 +- webrtc/video_engine/vie_encoder.cc | 29 ++++++------------- 3 files changed, 11 insertions(+), 24 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index 1a3b79cbc2..b4dfefcb36 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -608,9 +608,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) { if (!IsDefaultModule()) { // Don't send from default module. - if (SendingMedia()) { - return rtp_sender_.TimeToSendPadding(bytes); - } + return rtp_sender_.TimeToSendPadding(bytes); } else { CriticalSectionScoped lock(critical_section_module_ptrs_.get()); for (size_t i = 0; i < child_modules_.size(); ++i) { diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h index cad0e5ab87..ef9209a444 100644 --- a/webrtc/modules/video_coding/main/interface/video_coding.h +++ b/webrtc/modules/video_coding/main/interface/video_coding.h @@ -238,7 +238,7 @@ public: // frame rate/dimensions need to be updated for video quality optimization // // Input: - // - videoQMSettings : The callback object to register. + // - videoQMSettings : The callback object to register. // // Return value : VCM_OK, on success. // < 0, on error diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc index ba7862a81d..0955066b11 100644 --- a/webrtc/video_engine/vie_encoder.cc +++ b/webrtc/video_engine/vie_encoder.cc @@ -187,8 +187,13 @@ bool ViEEncoder::Init() { qm_callback_ = new QMVideoSettingsCallback(&vpm_); #ifdef VIDEOCODEC_VP8 + VideoCodecType codec_type = webrtc::kVideoCodecVP8; +#else + VideoCodecType codec_type = webrtc::kVideoCodecI420; +#endif + VideoCodec video_codec; - if (vcm_.Codec(webrtc::kVideoCodecVP8, &video_codec) != VCM_OK) { + if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) { return false; } { @@ -202,21 +207,6 @@ bool ViEEncoder::Init() { if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) { 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) { 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 // start and stop sending. - if (default_rtp_rtcp_->Sending() == false) { - if (default_rtp_rtcp_->SetSendingStatus(true) != 0) { - return -1; - } + if (default_rtp_rtcp_->SetSendingStatus(true) != 0) { + return -1; } + bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(), video_codec.startBitrate * 1000, video_codec.minBitrate * 1000,