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5eae1d994e
Bug: webrtc:9719 Change-Id: I04e892ce0f2af5c48040dd92ff0701209104fe65 Reviewed-on: https://webrtc-review.googlesource.com/c/111287 Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25734}
124 lines
4.4 KiB
C++
124 lines
4.4 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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MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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int sampling_rate_hz,
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int starting_sample_index,
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int samples_per_channel,
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int sequence_number,
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FrameType frame_type,
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uint8_t payload_type,
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std::vector<uint8_t> encoded_data)
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: sampling_rate_hz_(sampling_rate_hz),
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starting_sample_index_(starting_sample_index),
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samples_per_channel_(samples_per_channel),
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sequence_number_(sequence_number),
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frame_type_(frame_type),
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payload_type_(payload_type),
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encoded_data_(std::move(encoded_data)) {}
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MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
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const MediaTransportEncodedAudioFrame&) = default;
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MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
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MediaTransportEncodedAudioFrame&&) = default;
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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const MediaTransportEncodedAudioFrame&) = default;
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MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
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MediaTransportEncodedAudioFrame&&) = default;
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MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() {}
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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int64_t frame_id,
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std::vector<int64_t> referenced_frame_ids,
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VideoCodecType codec_type,
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const webrtc::EncodedImage& encoded_image)
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: codec_type_(codec_type),
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encoded_image_(encoded_image),
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frame_id_(frame_id),
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referenced_frame_ids_(std::move(referenced_frame_ids)) {}
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MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
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const MediaTransportEncodedVideoFrame&) = default;
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MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
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MediaTransportEncodedVideoFrame&&) = default;
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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const MediaTransportEncodedVideoFrame&) = default;
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MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
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MediaTransportEncodedVideoFrame&&) = default;
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SendDataParams::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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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::SetNetworkChangeCallback(
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MediaTransportNetworkChangeCallback* callback) {}
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void MediaTransportInterface::RemoveTargetTransferRateObserver(
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webrtc::TargetTransferRateObserver* observer) {}
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void MediaTransportInterface::AddTargetTransferRateObserver(
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webrtc::TargetTransferRateObserver* observer) {}
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size_t MediaTransportInterface::GetAudioPacketOverhead() const {
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return 0;
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}
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} // namespace webrtc
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