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Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.

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R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2013-12-12 22:40:39 +00:00
parent 451745ec05
commit a129b6cd13
101 changed files with 490 additions and 290 deletions
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8
talk/media/base/fakemediaengine.h
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@@ -191,10 +191,12 @@ template <class Base> class RtpHelper : public Base {
return true;
}
void set_playout(bool playout) { playout_ = playout; }
virtual void OnPacketReceived(talk_base::Buffer* packet) {
virtual void OnPacketReceived(talk_base::Buffer* packet,
const talk_base::PacketTime& packet_time) {
rtp_packets_.push_back(std::string(packet->data(), packet->length()));
}
virtual void OnRtcpReceived(talk_base::Buffer* packet) {
virtual void OnRtcpReceived(talk_base::Buffer* packet,
const talk_base::PacketTime& packet_time) {
rtcp_packets_.push_back(std::string(packet->data(), packet->length()));
}
virtual void OnReadyToSend(bool ready) {
@@ -776,6 +778,8 @@ class FakeVoiceEngine : public FakeBaseEngine {
bool SetLocalMonitor(bool enable) { return true; }
bool StartAecDump(FILE* file) { return false; }
bool RegisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
MediaProcessorDirection direction) {
if (direction == MPD_RX) {