| /* |
| * 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 |