| /* |
| * Copyright 2018 The WebRTC Project Authors. All rights reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_ |
| #define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_ |
| |
| #include <utility> |
| |
| #include "api/media_transport_interface.h" |
| |
| namespace webrtc { |
| |
| // Contains two MediaTransportsInterfaces that are connected to each other. |
| // Currently supports audio only. |
| class MediaTransportPair { |
| public: |
| MediaTransportPair() |
| : pipe_{LoopbackMediaTransport(&pipe_[1]), |
| LoopbackMediaTransport(&pipe_[0])} {} |
| |
| // Ownership stays with MediaTransportPair |
| MediaTransportInterface* first() { return &pipe_[0]; } |
| MediaTransportInterface* second() { return &pipe_[1]; } |
| |
| private: |
| class LoopbackMediaTransport : public MediaTransportInterface { |
| public: |
| explicit LoopbackMediaTransport(LoopbackMediaTransport* other) |
| : other_(other) {} |
| ~LoopbackMediaTransport() { RTC_CHECK(sink_ == nullptr); } |
| |
| RTCError SendAudioFrame(uint64_t channel_id, |
| MediaTransportEncodedAudioFrame frame) override { |
| other_->OnData(channel_id, std::move(frame)); |
| return RTCError::OK(); |
| }; |
| |
| RTCError SendVideoFrame( |
| uint64_t channel_id, |
| const MediaTransportEncodedVideoFrame& frame) override { |
| return RTCError::OK(); |
| } |
| |
| RTCError RequestKeyFrame(uint64_t channel_id) override { |
| return RTCError::OK(); |
| } |
| |
| void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override { |
| if (sink) { |
| RTC_CHECK(sink_ == nullptr); |
| } |
| sink_ = sink; |
| } |
| |
| void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {} |
| |
| void SetTargetTransferRateObserver( |
| webrtc::TargetTransferRateObserver* observer) override {} |
| |
| private: |
| void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame) { |
| if (sink_) { |
| sink_->OnData(channel_id, frame); |
| } |
| } |
| |
| MediaTransportAudioSinkInterface* sink_ = nullptr; |
| LoopbackMediaTransport* other_; |
| }; |
| |
| LoopbackMediaTransport pipe_[2]; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_ |