b9b146c9fe
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
105 lines
3.6 KiB
C++
105 lines
3.6 KiB
C++
/*
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* Copyright (c) 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 CALL_DEGRADED_CALL_H_
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#define CALL_DEGRADED_CALL_H_
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#include <memory>
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#include "absl/types/optional.h"
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#include "api/call/transport.h"
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#include "call/call.h"
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#include "call/fake_network_pipe.h"
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#include "modules/utility/include/process_thread.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class DegradedCall : public Call, private Transport, private PacketReceiver {
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public:
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explicit DegradedCall(std::unique_ptr<Call> call,
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absl::optional<FakeNetworkPipe::Config> send_config,
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absl::optional<FakeNetworkPipe::Config> receive_config);
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~DegradedCall() override;
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// Implements Call.
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AudioSendStream* CreateAudioSendStream(
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const AudioSendStream::Config& config) override;
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void DestroyAudioSendStream(AudioSendStream* send_stream) override;
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AudioReceiveStream* CreateAudioReceiveStream(
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const AudioReceiveStream::Config& config) override;
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void DestroyAudioReceiveStream(AudioReceiveStream* receive_stream) override;
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VideoSendStream* CreateVideoSendStream(
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config) override;
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VideoSendStream* CreateVideoSendStream(
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config,
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std::unique_ptr<FecController> fec_controller) override;
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void DestroyVideoSendStream(VideoSendStream* send_stream) override;
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VideoReceiveStream* CreateVideoReceiveStream(
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VideoReceiveStream::Config configuration) override;
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void DestroyVideoReceiveStream(VideoReceiveStream* receive_stream) override;
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FlexfecReceiveStream* CreateFlexfecReceiveStream(
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const FlexfecReceiveStream::Config& config) override;
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void DestroyFlexfecReceiveStream(
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FlexfecReceiveStream* receive_stream) override;
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PacketReceiver* Receiver() override;
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RtpTransportControllerSendInterface* GetTransportControllerSend() override;
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Stats GetStats() const override;
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void SetBitrateAllocationStrategy(
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std::unique_ptr<rtc::BitrateAllocationStrategy>
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bitrate_allocation_strategy) override;
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void SignalChannelNetworkState(MediaType media, NetworkState state) override;
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void OnTransportOverheadChanged(MediaType media,
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int transport_overhead_per_packet) override;
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void OnSentPacket(const rtc::SentPacket& sent_packet) override;
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protected:
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// Implements Transport.
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bool SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) override;
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bool SendRtcp(const uint8_t* packet, size_t length) override;
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// Implements PacketReceiver.
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DeliveryStatus DeliverPacket(MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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const PacketTime& packet_time) override;
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private:
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Clock* const clock_;
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const std::unique_ptr<Call> call_;
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const absl::optional<FakeNetworkPipe::Config> send_config_;
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const std::unique_ptr<ProcessThread> send_process_thread_;
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std::unique_ptr<FakeNetworkPipe> send_pipe_;
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size_t num_send_streams_;
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const absl::optional<FakeNetworkPipe::Config> receive_config_;
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std::unique_ptr<FakeNetworkPipe> receive_pipe_;
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};
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} // namespace webrtc
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#endif // CALL_DEGRADED_CALL_H_
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