From 64e2cbf184ecf8f20fb949ea5a1c6e1c1bdd8bc3 Mon Sep 17 00:00:00 2001 From: "tnakamura@webrtc.org" Date: Tue, 16 Jul 2013 21:52:59 +0000 Subject: [PATCH] clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos Reason for recent series of reverts: video freezes when testing with packet loss R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1817004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../modules/rtp_rtcp/source/rtp_payload_registry.cc | 1 - .../modules/rtp_rtcp/test/testAPI/test_api_video.cc | 1 - webrtc/voice_engine/channel.cc | 11 ----------- 3 files changed, 13 deletions(-) diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index 39cea6b523..cb689ee424 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -283,7 +283,6 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy { ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload; payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); - assert(frequency >= 1000); payload->typeSpecific.Audio.frequency = frequency; payload->typeSpecific.Audio.channels = channels; payload->typeSpecific.Audio.rate = rate; diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc index f8c786ab1e..e2c2eff50e 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc @@ -15,7 +15,6 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h" diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 9d76761181..ba84ba9d73 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -2151,17 +2151,6 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) { "IncomingPacket invalid RTP header"); return -1; } - header.payload_type_frequency = - rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType); - bool retransmitted = IsPacketRetransmitted(header); - bool in_order = rtp_receiver_->InOrderPacket(header.sequenceNumber); - rtp_receive_statistics_->IncomingPacket(header, static_cast(length), - retransmitted, in_order); - PayloadUnion payload_specific; - if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType, - &payload_specific)) { - return -1; - } // Deliver RTP packet to RTP/RTCP module for parsing // The packet will be pushed back to the channel thru the // OnReceivedPayloadData callback so we don't push it to the ACM here