From 68f2168978548bb8dd80ba77a888722bf9e5ec66 Mon Sep 17 00:00:00 2001 From: "perkj@webrtc.org" Date: Wed, 30 Nov 2011 18:11:23 +0000 Subject: [PATCH] Remove global voe::Channel::numSocketThreads. git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/voice_engine/main/source/channel.cc | 5 ++--- src/voice_engine/main/source/channel.h | 2 +- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc index a4b65eef6e..d5490a8979 100644 --- a/src/voice_engine/main/source/channel.cc +++ b/src/voice_engine/main/source/channel.cc @@ -976,8 +976,6 @@ Channel::NeededFrequency(const WebRtc_Word32 id) return(highestNeeded); } -WebRtc_UWord8 Channel::numSocketThreads = KNumSocketThreads; - WebRtc_Word32 Channel::CreateChannel(Channel*& channel, const WebRtc_Word32 channelId, @@ -1078,8 +1076,9 @@ Channel::Channel(const WebRtc_Word32 channelId, _audioCodingModule(*AudioCodingModule::Create( VoEModuleId(instanceId, channelId))), #ifndef WEBRTC_EXTERNAL_TRANSPORT + _numSocketThreads(KNumSocketThreads), _socketTransportModule(*UdpTransport::Create( - VoEModuleId(instanceId, channelId), numSocketThreads)), + VoEModuleId(instanceId, channelId), _numSocketThreads)), #endif #ifdef WEBRTC_SRTP _srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId, diff --git a/src/voice_engine/main/source/channel.h b/src/voice_engine/main/source/channel.h index e18ba77bdb..c9d5b829bd 100644 --- a/src/voice_engine/main/source/channel.h +++ b/src/voice_engine/main/source/channel.h @@ -79,7 +79,6 @@ class Channel: public: enum {KNumSocketThreads = 1}; enum {KNumberOfSocketBuffers = 8}; - static WebRtc_UWord8 numSocketThreads; public: virtual ~Channel(); static WebRtc_Word32 CreateChannel(Channel*& channel, @@ -543,6 +542,7 @@ private: RtpRtcp& _rtpRtcpModule; AudioCodingModule& _audioCodingModule; #ifndef WEBRTC_EXTERNAL_TRANSPORT + WebRtc_UWord8 _numSocketThreads; UdpTransport& _socketTransportModule; #endif #ifdef WEBRTC_SRTP