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Delete telephone-event handling from RTPReceiverAudio.

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Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}
This commit is contained in:
Niels Möller
2018-07-30 16:10:41 +02:00
committed by Commit Bot
parent 31f1596c7c
commit ab4a530b87
7 changed files with 2 additions and 89 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
audio/channel.cc
View File

@ -529,7 +529,6 @@ Channel::Channel(ProcessThread* module_process_thread,
RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
this,
rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
_outputAudioLevel(),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
@ -616,7 +615,6 @@ void Channel::Init() {
// disabled by the user.
// After StopListen (when no sockets exists), RTCP packets will no longer
// be transmitted since the Transport object will then be invalid.
telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
// RTCP is enabled by default.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);

2
audio/channel.h
View File

@ -56,7 +56,6 @@ class RTPReceiverAudio;
class RtpPacketReceived;
class RtpRtcp;
class RtpTransportControllerSendInterface;
class TelephoneEventHandler;
struct SenderInfo;
@ -342,7 +341,6 @@ class Channel
std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
std::unique_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
std::unique_ptr<RtpRtcp> _rtpRtcpModule;
std::unique_ptr<AudioCodingModule> audio_coding_;
AudioSinkInterface* audio_sink_ = nullptr;