chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc

/*
 *  Copyright (c) 2020 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.
 */

#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"

#include <cstddef>
#include <cstdint>
#include <memory>
#include <string>
#include <utility>
#include <vector>

#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/frame_transformer_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_frame_metadata.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_layers_allocation.h"
#include "api/video_codecs/video_codec.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"

namespace webrtc {
namespace {

// Using a reasonable default of 10ms for the retransmission delay for frames
// not coming from this sender's encoder. This is usually taken from an
// estimate of the RTT of the link,so 10ms should be a reasonable estimate for
// frames being re-transmitted to a peer, probably on the same network.
const TimeDelta kDefaultRetransmissionsTime = …;
}  // namespace

class TransformableVideoSenderFrame : public TransformableVideoFrameInterface { … };

RTPSenderVideoFrameTransformerDelegate::RTPSenderVideoFrameTransformerDelegate(
    RTPVideoFrameSenderInterface* sender,
    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
    uint32_t ssrc,
    TaskQueueFactory* task_queue_factory)
    : … { … }

void RTPSenderVideoFrameTransformerDelegate::Init() { … }

bool RTPSenderVideoFrameTransformerDelegate::TransformFrame(
    int payload_type,
    absl::optional<VideoCodecType> codec_type,
    uint32_t rtp_timestamp,
    const EncodedImage& encoded_image,
    RTPVideoHeader video_header,
    TimeDelta expected_retransmission_time) { … }

void RTPSenderVideoFrameTransformerDelegate::OnTransformedFrame(
    std::unique_ptr<TransformableFrameInterface> frame) { … }

void RTPSenderVideoFrameTransformerDelegate::StartShortCircuiting() { … }

void RTPSenderVideoFrameTransformerDelegate::SendVideo(
    std::unique_ptr<TransformableFrameInterface> transformed_frame) const { … }

void RTPSenderVideoFrameTransformerDelegate::SetVideoStructureUnderLock(
    const FrameDependencyStructure* video_structure) { … }

void RTPSenderVideoFrameTransformerDelegate::SetVideoLayersAllocationUnderLock(
    VideoLayersAllocation allocation) { … }

void RTPSenderVideoFrameTransformerDelegate::Reset() { … }

std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
    TransformableVideoFrameInterface* original) { … }

}  // namespace webrtc