eb2c6415a9
This change deletes the default implementations of state and data channel methods (SetMediaTransportStateCallback, SendData, CloseChannel, and SetDataSink). It adds stub implementations to LoopbackMediaTransport and FakeMediaTransport. Bug: webrtc:9719 Change-Id: I49b7780c055b552330546b460c2e79ce8df81833 Reviewed-on: https://webrtc-review.googlesource.com/c/108940 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25457}
99 lines
2.9 KiB
C++
99 lines
2.9 KiB
C++
/*
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* Copyright 2018 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_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
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#define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
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#include <utility>
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#include "api/media_transport_interface.h"
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namespace webrtc {
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// Contains two MediaTransportsInterfaces that are connected to each other.
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// Currently supports audio only.
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class MediaTransportPair {
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public:
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MediaTransportPair()
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: pipe_{LoopbackMediaTransport(&pipe_[1]),
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LoopbackMediaTransport(&pipe_[0])} {}
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// Ownership stays with MediaTransportPair
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MediaTransportInterface* first() { return &pipe_[0]; }
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MediaTransportInterface* second() { return &pipe_[1]; }
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private:
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class LoopbackMediaTransport : public MediaTransportInterface {
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public:
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explicit LoopbackMediaTransport(LoopbackMediaTransport* other)
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: other_(other) {}
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~LoopbackMediaTransport() { RTC_CHECK(sink_ == nullptr); }
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RTCError SendAudioFrame(uint64_t channel_id,
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MediaTransportEncodedAudioFrame frame) override {
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other_->OnData(channel_id, std::move(frame));
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return RTCError::OK();
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};
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RTCError SendVideoFrame(
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uint64_t channel_id,
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const MediaTransportEncodedVideoFrame& frame) override {
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return RTCError::OK();
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}
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RTCError RequestKeyFrame(uint64_t channel_id) override {
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return RTCError::OK();
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}
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void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {
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if (sink) {
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RTC_CHECK(sink_ == nullptr);
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}
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sink_ = sink;
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}
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void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {}
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void SetTargetTransferRateObserver(
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webrtc::TargetTransferRateObserver* observer) override {}
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void SetMediaTransportStateCallback(
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MediaTransportStateCallback* callback) override {}
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RTCError SendData(int channel_id,
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& buffer) override {
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return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
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}
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RTCError CloseChannel(int channel_id) override {
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return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
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}
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void SetDataSink(DataChannelSink* sink) override {}
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private:
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void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame) {
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if (sink_) {
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sink_->OnData(channel_id, frame);
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}
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}
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MediaTransportAudioSinkInterface* sink_ = nullptr;
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LoopbackMediaTransport* other_;
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};
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LoopbackMediaTransport pipe_[2];
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};
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} // namespace webrtc
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#endif // API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
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