blob: 039a4a1e57e4686d961473bb488c956fe27e9ee8 [file] [log] [blame]
/*
* 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;
MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
int sampling_rate_hz,
int starting_sample_index,
int samples_per_channel,
int sequence_number,
FrameType frame_type,
uint8_t payload_type,
std::vector<uint8_t> encoded_data)
: sampling_rate_hz_(sampling_rate_hz),
starting_sample_index_(starting_sample_index),
samples_per_channel_(samples_per_channel),
sequence_number_(sequence_number),
frame_type_(frame_type),
payload_type_(payload_type),
encoded_data_(std::move(encoded_data)) {}
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
const MediaTransportEncodedAudioFrame&) = default;
MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
MediaTransportEncodedAudioFrame&&) = default;
MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() {}
MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
int64_t frame_id,
std::vector<int64_t> referenced_frame_ids,
VideoCodecType codec_type,
const webrtc::EncodedImage& encoded_image)
: codec_type_(codec_type),
encoded_image_(encoded_image),
frame_id_(frame_id),
referenced_frame_ids_(std::move(referenced_frame_ids)) {}
MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
const MediaTransportEncodedVideoFrame&) = default;
MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
MediaTransportEncodedVideoFrame&&) = default;
MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
const MediaTransportEncodedVideoFrame&) = default;
MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
MediaTransportEncodedVideoFrame&&) = default;
SendDataParams::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);
}
absl::optional<TargetTransferRate>
MediaTransportInterface::GetLatestTargetTransferRate() {
return absl::nullopt;
}
void MediaTransportInterface::SetNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback) {}
void MediaTransportInterface::RemoveTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer) {}
void MediaTransportInterface::SetTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer) {}
void MediaTransportInterface::AddTargetTransferRateObserver(
webrtc::TargetTransferRateObserver* observer) {}
size_t MediaTransportInterface::GetAudioPacketOverhead() const {
return 0;
}
} // namespace webrtc