Fix compilation issues on media_transport_interface.h
Include api/video/encoded_image.h, and move constructors and destructors to .cc file. Bug: webrtc:9719 Change-Id: Ibecdc1151bf672155d3c09e13749ac39c921c3aa Reviewed-on: https://webrtc-review.googlesource.com/c/104560 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25044}
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@ -63,6 +63,7 @@ rtc_static_library("libjingle_peerconnection_api") {
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"jsepicecandidate.cc",
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"jsepicecandidate.h",
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"jsepsessiondescription.h",
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"media_transport_interface.cc",
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"media_transport_interface.h",
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"mediaconstraintsinterface.cc",
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"mediaconstraintsinterface.h",
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@ -109,6 +110,7 @@ rtc_static_library("libjingle_peerconnection_api") {
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"audio_codecs:audio_codecs_api",
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"transport:bitrate_settings",
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"transport:network_control",
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"video:encoded_image",
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"video:video_frame",
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"//third_party/abseil-cpp/absl/types:optional",
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52
api/media_transport_interface.cc
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52
api/media_transport_interface.cc
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@ -0,0 +1,52 @@
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/*
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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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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() = default;
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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() = default;
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} // namespace webrtc
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@ -23,6 +23,7 @@
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#include <vector>
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#include "api/rtcerror.h"
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#include "api/video/encoded_image.h"
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#include "common_types.h" // NOLINT(build/include)
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namespace rtc {
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@ -75,14 +76,9 @@ class MediaTransportEncodedAudioFrame {
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uint8_t payload_type,
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// Vector with opaque encoded data.
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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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std::vector<uint8_t> encoded_data);
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~MediaTransportEncodedAudioFrame();
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// Getters.
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int sampling_rate_hz() const { return sampling_rate_hz_; }
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@ -130,11 +126,8 @@ class MediaTransportEncodedVideoFrame {
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MediaTransportEncodedVideoFrame(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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const webrtc::EncodedImage& encoded_image);
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~MediaTransportEncodedVideoFrame();
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VideoCodecType codec_type() const { return codec_type_; }
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const webrtc::EncodedImage& encoded_image() const { return encoded_image_; }
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