webrtc/api/media_transport_interface.cc
Bjorn Mellem f58e43e2a6 Add an OpenChannel method to MediaTransportInterface and call it whenever PeerConnection opens a new data channel.
This informs the media transport that PeerConnection wants to use a data channel
and gives it a chance to set up before the data channel sends the first message.

Bug: webrtc:9719
Change-Id: I6ea905a74b29b8735e77ac68bc8606e7bca77f18
Reviewed-on: https://webrtc-review.googlesource.com/c/124020
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26823}
2019-02-22 20:55:12 +00:00

98 lines
3.4 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>
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,
bool is_caller) {
MediaTransportSettings settings;
settings.is_caller = is_caller;
return CreateMediaTransport(packet_transport, network_thread, settings);
}
RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
MediaTransportFactory::CreateMediaTransport(
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
return std::unique_ptr<MediaTransportInterface>(nullptr);
}
MediaTransportInterface::MediaTransportInterface() = default;
MediaTransportInterface::~MediaTransportInterface() = default;
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) {}
// TODO(mellem): Delete when all implementations support it.
RTCError MediaTransportInterface::OpenChannel(int channel_id) {
// NB: This must return OK to avoid breaking existing implementations, which
// do not require calling OpenChannel.
return RTCError::OK();
}
} // namespace webrtc