mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00
a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
108 lines
3.6 KiB
C++
108 lines
3.6 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
// This is EXPERIMENTAL interface for media transport.
|
|
//
|
|
// The goal is to refactor WebRTC code so that audio and video frames
|
|
// are sent / received through the media transport interface. This will
|
|
// enable different media transport implementations, including QUIC-based
|
|
// media transport.
|
|
|
|
#include "api/media_transport_interface.h"
|
|
|
|
#include <cstdint>
|
|
#include <utility>
|
|
|
|
#include "api/datagram_transport_interface.h"
|
|
|
|
namespace webrtc {
|
|
|
|
MediaTransportSettings::MediaTransportSettings() = default;
|
|
MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
|
|
default;
|
|
MediaTransportSettings& MediaTransportSettings::operator=(
|
|
const MediaTransportSettings&) = default;
|
|
MediaTransportSettings::~MediaTransportSettings() = default;
|
|
|
|
SendDataParams::SendDataParams() = default;
|
|
SendDataParams::SendDataParams(const SendDataParams&) = default;
|
|
|
|
RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
|
|
MediaTransportFactory::CreateMediaTransport(
|
|
rtc::PacketTransportInternal* packet_transport,
|
|
rtc::Thread* network_thread,
|
|
const MediaTransportSettings& settings) {
|
|
return std::unique_ptr<MediaTransportInterface>(nullptr);
|
|
}
|
|
|
|
RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
|
|
MediaTransportFactory::CreateMediaTransport(
|
|
rtc::Thread* network_thread,
|
|
const MediaTransportSettings& settings) {
|
|
return std::unique_ptr<MediaTransportInterface>(nullptr);
|
|
}
|
|
|
|
RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
|
|
MediaTransportFactory::CreateDatagramTransport(
|
|
rtc::Thread* network_thread,
|
|
const MediaTransportSettings& settings) {
|
|
return std::unique_ptr<DatagramTransportInterface>(nullptr);
|
|
}
|
|
|
|
std::string MediaTransportFactory::GetTransportName() const {
|
|
return "";
|
|
}
|
|
|
|
MediaTransportInterface::MediaTransportInterface() = default;
|
|
MediaTransportInterface::~MediaTransportInterface() = default;
|
|
|
|
absl::optional<std::string>
|
|
MediaTransportInterface::GetTransportParametersOffer() const {
|
|
return absl::nullopt;
|
|
}
|
|
|
|
void MediaTransportInterface::Connect(
|
|
rtc::PacketTransportInternal* packet_transport) {}
|
|
|
|
void MediaTransportInterface::SetKeyFrameRequestCallback(
|
|
MediaTransportKeyFrameRequestCallback* callback) {}
|
|
|
|
absl::optional<TargetTransferRate>
|
|
MediaTransportInterface::GetLatestTargetTransferRate() {
|
|
return absl::nullopt;
|
|
}
|
|
|
|
void MediaTransportInterface::AddNetworkChangeCallback(
|
|
MediaTransportNetworkChangeCallback* callback) {}
|
|
|
|
void MediaTransportInterface::RemoveNetworkChangeCallback(
|
|
MediaTransportNetworkChangeCallback* callback) {}
|
|
|
|
void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
|
|
AudioPacketReceivedObserver* observer) {}
|
|
|
|
void MediaTransportInterface::AddTargetTransferRateObserver(
|
|
TargetTransferRateObserver* observer) {}
|
|
void MediaTransportInterface::RemoveTargetTransferRateObserver(
|
|
TargetTransferRateObserver* observer) {}
|
|
|
|
void MediaTransportInterface::AddRttObserver(
|
|
MediaTransportRttObserver* observer) {}
|
|
void MediaTransportInterface::RemoveRttObserver(
|
|
MediaTransportRttObserver* observer) {}
|
|
|
|
size_t MediaTransportInterface::GetAudioPacketOverhead() const {
|
|
return 0;
|
|
}
|
|
|
|
void MediaTransportInterface::SetAllocatedBitrateLimits(
|
|
const MediaTransportAllocatedBitrateLimits& limits) {}
|
|
|
|
} // namespace webrtc
|