04f4d126f8
Timing information is gathered in EncodedImage, starting at encoders. Then it's sent using RTP header extension. In the end, it's gathered at the GenericDecoder. Actual reporting and tests will be in the next CLs. BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/2911193002 Cr-Commit-Position: refs/heads/master@{#18659}
90 lines
3.2 KiB
C++
90 lines
3.2 KiB
C++
/*
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* Copyright (c) 2014 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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#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
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#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
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// TODO(nisse): This header file should eventually be deleted. The
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// EncodedImage class stays in this file until we have figured out how
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// to refactor and clean up related interfaces, at which point it
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// should be moved to somewhere under api/.
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#include "webrtc/common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// TODO(pbos): Rename EncodedFrame and reformat this class' members.
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class EncodedImage {
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public:
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static const size_t kBufferPaddingBytesH264;
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// Some decoders require encoded image buffers to be padded with a small
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// number of additional bytes (due to over-reading byte readers).
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static size_t GetBufferPaddingBytes(VideoCodecType codec_type);
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EncodedImage() : EncodedImage(nullptr, 0, 0) {}
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EncodedImage(uint8_t* buffer, size_t length, size_t size)
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: _buffer(buffer), _length(length), _size(size) {}
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void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms) const {
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timing_.is_timing_frame = true;
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timing_.encode_start_ms = encode_start_ms;
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timing_.encode_finish_ms = encode_finish_ms;
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}
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// TODO(kthelgason): get rid of this struct as it only has a single member
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// remaining.
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struct AdaptReason {
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AdaptReason() : bw_resolutions_disabled(-1) {}
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int bw_resolutions_disabled; // Number of resolutions that are not sent
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// due to bandwidth for this frame.
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// Or -1 if information is not provided.
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};
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uint32_t _encodedWidth = 0;
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uint32_t _encodedHeight = 0;
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uint32_t _timeStamp = 0;
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// NTP time of the capture time in local timebase in milliseconds.
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int64_t ntp_time_ms_ = 0;
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int64_t capture_time_ms_ = 0;
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FrameType _frameType = kVideoFrameDelta;
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uint8_t* _buffer;
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size_t _length;
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size_t _size;
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VideoRotation rotation_ = kVideoRotation_0;
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VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
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bool _completeFrame = false;
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AdaptReason adapt_reason_;
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int qp_ = -1; // Quantizer value.
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// When an application indicates non-zero values here, it is taken as an
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// indication that all future frames will be constrained with those limits
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// until the application indicates a change again.
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PlayoutDelay playout_delay_ = {-1, -1};
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// Timing information should be updatable on const instances.
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mutable struct Timing {
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bool is_timing_frame = false;
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int64_t encode_start_ms = 0;
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int64_t encode_finish_ms = 0;
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int64_t packetization_finish_ms = 0;
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int64_t pacer_exit_ms = 0;
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int64_t network_timestamp_ms = 0;
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int64_t network2_timestamp_ms = 0;
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int64_t receive_start_ms = 0;
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int64_t receive_finish_ms = 0;
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} timing_;
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
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