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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
79 lines
2.8 KiB
C++
79 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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#include <cstdint>
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#include <utility>
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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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