mirror of
https://github.com/mollyim/webrtc.git
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990d6b875e
This reverts commit 90bace0958.
Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.
Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
>
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
>
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
>
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
>
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
>
> TBR=solenberg
>
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}
TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org
Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
119 lines
4.8 KiB
C++
119 lines
4.8 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_PEERCONNECTIONPROXY_H_
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#define API_PEERCONNECTIONPROXY_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/peerconnectioninterface.h"
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#include "api/proxy.h"
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namespace webrtc {
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// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
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// are called on is an implementation detail.
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BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, local_streams)
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PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, remote_streams)
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PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
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PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
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PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
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AddTrack,
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MediaStreamTrackInterface*,
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std::vector<MediaStreamInterface*>)
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PROXY_METHOD1(bool, RemoveTrack, RtpSenderInterface*)
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PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
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CreateDtmfSender,
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AudioTrackInterface*)
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PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
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CreateSender,
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const std::string&,
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const std::string&)
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PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
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GetSenders)
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PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
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GetReceivers)
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PROXY_METHOD3(bool,
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GetStats,
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StatsObserver*,
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MediaStreamTrackInterface*,
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StatsOutputLevel)
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PROXY_METHOD1(void, GetStats, RTCStatsCollectorCallback*)
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PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
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CreateDataChannel,
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const std::string&,
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const DataChannelInit*)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
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pending_local_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
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pending_remote_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
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current_local_description)
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PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
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current_remote_description)
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PROXY_METHOD2(void,
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CreateOffer,
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CreateSessionDescriptionObserver*,
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const MediaConstraintsInterface*)
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PROXY_METHOD2(void,
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CreateAnswer,
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CreateSessionDescriptionObserver*,
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const MediaConstraintsInterface*)
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PROXY_METHOD2(void,
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CreateOffer,
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CreateSessionDescriptionObserver*,
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const RTCOfferAnswerOptions&)
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PROXY_METHOD2(void,
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CreateAnswer,
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CreateSessionDescriptionObserver*,
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const RTCOfferAnswerOptions&)
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PROXY_METHOD2(void,
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SetLocalDescription,
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SetSessionDescriptionObserver*,
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SessionDescriptionInterface*)
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PROXY_METHOD2(void,
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SetRemoteDescription,
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SetSessionDescriptionObserver*,
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SessionDescriptionInterface*)
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PROXY_METHOD0(PeerConnectionInterface::RTCConfiguration, GetConfiguration);
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PROXY_METHOD2(bool,
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SetConfiguration,
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const PeerConnectionInterface::RTCConfiguration&,
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RTCError*);
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PROXY_METHOD1(bool,
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SetConfiguration,
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const PeerConnectionInterface::RTCConfiguration&);
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PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*)
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PROXY_METHOD1(bool,
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RemoveIceCandidates,
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const std::vector<cricket::Candidate>&);
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PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
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PROXY_METHOD1(RTCError, SetBitrate, const BitrateParameters&);
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PROXY_METHOD1(void,
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SetBitrateAllocationStrategy,
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std::unique_ptr<rtc::BitrateAllocationStrategy>);
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PROXY_METHOD0(SignalingState, signaling_state)
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PROXY_METHOD0(IceConnectionState, ice_connection_state)
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PROXY_METHOD0(IceGatheringState, ice_gathering_state)
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PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
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PROXY_METHOD1(bool, StartRtcEventLog, std::unique_ptr<RtcEventLogOutput>)
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PROXY_METHOD0(void, StopRtcEventLog)
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PROXY_METHOD0(void, Close)
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END_PROXY_MAP()
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} // namespace webrtc
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#endif // API_PEERCONNECTIONPROXY_H_
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