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e049eba27c
This reverts commit 0d8eed6ac7.
Reason for revert: crashes of unit tests.
Original change's description:
> Add Sender and Receiver interfaces for MediaTransport audio
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> Implement in LoopbackMediaTransport.
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> Bug: webrtc:9719
> Change-Id: I429ac3f78d99b8ea4f9ac85b9a3600b215b61a55
> Reviewed-on: https://webrtc-review.googlesource.com/c/121957
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26731}
TBR=solenberg@webrtc.org,nisse@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
Change-Id: I02e409e1bbe2b2dea8a7b1aa08fa44d4146bda8f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/123232
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26733}
91 lines
3.1 KiB
C++
91 lines
3.1 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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// This is EXPERIMENTAL interface for media transport.
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//
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// The goal is to refactor WebRTC code so that audio and video frames
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// are sent / received through the media transport interface. This will
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// enable different media transport implementations, including QUIC-based
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// media transport.
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#include "api/media_transport_interface.h"
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#include <cstdint>
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#include <utility>
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namespace webrtc {
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MediaTransportSettings::MediaTransportSettings() = default;
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MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
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default;
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MediaTransportSettings& MediaTransportSettings::operator=(
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const MediaTransportSettings&) = default;
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MediaTransportSettings::~MediaTransportSettings() = default;
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SendDataParams::SendDataParams() = default;
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SendDataParams::SendDataParams(const SendDataParams&) = default;
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RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
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MediaTransportFactory::CreateMediaTransport(
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rtc::PacketTransportInternal* packet_transport,
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rtc::Thread* network_thread,
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bool is_caller) {
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MediaTransportSettings settings;
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settings.is_caller = is_caller;
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return CreateMediaTransport(packet_transport, network_thread, settings);
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}
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RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
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MediaTransportFactory::CreateMediaTransport(
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rtc::PacketTransportInternal* packet_transport,
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rtc::Thread* network_thread,
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const MediaTransportSettings& settings) {
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return std::unique_ptr<MediaTransportInterface>(nullptr);
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}
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MediaTransportInterface::MediaTransportInterface() = default;
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MediaTransportInterface::~MediaTransportInterface() = default;
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void MediaTransportInterface::SetKeyFrameRequestCallback(
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MediaTransportKeyFrameRequestCallback* callback) {}
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absl::optional<TargetTransferRate>
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MediaTransportInterface::GetLatestTargetTransferRate() {
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return absl::nullopt;
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}
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void MediaTransportInterface::AddNetworkChangeCallback(
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MediaTransportNetworkChangeCallback* callback) {}
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void MediaTransportInterface::RemoveNetworkChangeCallback(
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MediaTransportNetworkChangeCallback* callback) {}
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void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
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AudioPacketReceivedObserver* observer) {}
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void MediaTransportInterface::AddTargetTransferRateObserver(
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TargetTransferRateObserver* observer) {}
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void MediaTransportInterface::RemoveTargetTransferRateObserver(
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TargetTransferRateObserver* observer) {}
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void MediaTransportInterface::AddRttObserver(
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MediaTransportRttObserver* observer) {}
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void MediaTransportInterface::RemoveRttObserver(
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MediaTransportRttObserver* observer) {}
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size_t MediaTransportInterface::GetAudioPacketOverhead() const {
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return 0;
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}
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void MediaTransportInterface::SetAllocatedBitrateLimits(
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const MediaTransportAllocatedBitrateLimits& limits) {}
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} // namespace webrtc
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