From 6b34124a6d4733cf2d95986b4920bccec8b39397 Mon Sep 17 00:00:00 2001 From: solenberg Date: Mon, 6 Feb 2017 13:39:38 -0800 Subject: [PATCH] Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ BUG=webrtc:6847 Review-Url: https://codereview.webrtc.org/2663063008 Cr-Commit-Position: refs/heads/master@{#16457} --- webrtc/audio/audio_receive_stream.cc | 15 +-------------- webrtc/audio/audio_receive_stream.h | 2 -- webrtc/call/call.cc | 2 +- 3 files changed, 2 insertions(+), 17 deletions(-) diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 5cca45e2c1..8228a600b5 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -69,14 +69,12 @@ AudioReceiveStream::AudioReceiveStream( webrtc::RtcEventLog* event_log) : remote_bitrate_estimator_(remote_bitrate_estimator), config_(config), - audio_state_(audio_state), - rtp_header_parser_(RtpHeaderParser::Create()) { + audio_state_(audio_state) { LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); RTC_DCHECK_NE(config_.voe_channel_id, -1); RTC_DCHECK(audio_state_.get()); RTC_DCHECK(packet_router); RTC_DCHECK(remote_bitrate_estimator); - RTC_DCHECK(rtp_header_parser_); module_process_thread_checker_.DetachFromThread(); @@ -107,14 +105,8 @@ AudioReceiveStream::AudioReceiveStream( for (const auto& extension : config.rtp.extensions) { if (extension.uri == RtpExtension::kAudioLevelUri) { channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id); - bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( - kRtpExtensionAudioLevel, extension.id); - RTC_DCHECK(registered); } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) { channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id); - bool registered = rtp_header_parser_->RegisterRtpHeaderExtension( - kRtpExtensionTransportSequenceNumber, extension.id); - RTC_DCHECK(registered); } else { RTC_NOTREACHED() << "Unsupported RTP extension."; } @@ -321,11 +313,6 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, // calls on the worker thread. We should move towards always using a network // thread. Then this check can be enabled. // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); - RTPHeader header; - if (!rtp_header_parser_->Parse(packet, length, &header)) { - return false; - } - return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); } diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index 6721c7ee65..13869c453e 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -19,7 +19,6 @@ #include "webrtc/base/thread_checker.h" #include "webrtc/call/audio_receive_stream.h" #include "webrtc/call/syncable.h" -#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" namespace webrtc { class PacketRouter; @@ -81,7 +80,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, RemoteBitrateEstimator* const remote_bitrate_estimator_; const webrtc::AudioReceiveStream::Config config_; rtc::scoped_refptr audio_state_; - std::unique_ptr rtp_header_parser_; std::unique_ptr channel_proxy_; bool playing_ ACCESS_ON(worker_thread_checker_) = false; diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 6fd0f6d74a..5e21fe5312 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -208,7 +208,7 @@ class Call : public webrtc::Call, // extensions per SSRC instead, which leads to some storage overhead. RtpHeaderExtensionMap extensions; // Set if the RTCP feedback message needed for send side BWE was negotiated. - bool transport_cc; + bool transport_cc = false; }; std::map receive_rtp_config_ GUARDED_BY(receive_crit_);