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e0c2e97474
And use RTCConfiguration to enable/disable it on a per connection basis. With the advent of MediaTransportInterface, we need to be able to enable it on the per PeerConnection basis. At this point PeerConnection will not take any action when the MediaTransportInterface is set; this code will land a bit later, and will be accompanied by the tests that verify correct setup (hence no tests right now). At this point this is just a method stub to enable further development. Bug: webrtc:9719 Change-Id: I1f77d650cb03bf1191aa0b35669cd32f1b68446f Reviewed-on: https://webrtc-review.googlesource.com/c/103860 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25053}
76 lines
2.8 KiB
C++
76 lines
2.8 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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namespace webrtc {
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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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} // namespace webrtc
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