4b1afbe60a
This reverts commit c9a2c5e93aa51606916e6728454bcff26bb75f79. Reason for revert: Breaks downstream test Original change's description: > Reland "Copy video frames metadata between encoded and plain frames in one place" > > Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders. > > Currently some video frames metadata like rotation or ntp timestamps are > copied in every encoder and decoder separately. This CL makes copying to > happen at a single place for send or receive side. This will make it > easier to add new metadata in the future. > > Also, added some missing tests. > > Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 > > Bug: webrtc:10460 > Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102 > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27756} TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org Change-Id: I34cc563ec6383735c2a76a6f45a72a7726b74421 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10460 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134204 Reviewed-by: Artem Titarenko <artit@google.com> Commit-Queue: Artem Titarenko <artit@google.com> Cr-Commit-Position: refs/heads/master@{#27765}
85 lines
3.0 KiB
C++
85 lines
3.0 KiB
C++
/*
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* Copyright (c) 2019 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 VIDEO_FRAME_ENCODE_TIMER_H_
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#define VIDEO_FRAME_ENCODE_TIMER_H_
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#include <list>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/video/encoded_image.h"
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#include "api/video_codecs/video_codec.h"
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#include "api/video_codecs/video_encoder.h"
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#include "rtc_base/critical_section.h"
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namespace webrtc {
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class FrameEncodeTimer {
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public:
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explicit FrameEncodeTimer(EncodedImageCallback* frame_drop_callback);
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~FrameEncodeTimer();
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void OnEncoderInit(const VideoCodec& codec, bool internal_source);
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void OnSetRates(const VideoBitrateAllocation& bitrate_allocation,
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uint32_t framerate_fps);
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void OnEncodeStarted(uint32_t rtp_timestamp, int64_t capture_time_ms);
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void FillTimingInfo(size_t simulcast_svc_idx,
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EncodedImage* encoded_image,
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int64_t encode_done_ms);
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void Reset();
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private:
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size_t NumSpatialLayers() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
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// For non-internal-source encoders, returns encode started time and fixes
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// capture timestamp for the frame, if corrupted by the encoder.
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absl::optional<int64_t> ExtractEncodeStartTime(size_t simulcast_svc_idx,
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EncodedImage* encoded_image)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
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struct EncodeStartTimeRecord {
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EncodeStartTimeRecord(uint32_t timestamp,
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int64_t capture_time,
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int64_t encode_start_time)
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: rtp_timestamp(timestamp),
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capture_time_ms(capture_time),
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encode_start_time_ms(encode_start_time) {}
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uint32_t rtp_timestamp;
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int64_t capture_time_ms;
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int64_t encode_start_time_ms;
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};
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struct TimingFramesLayerInfo {
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TimingFramesLayerInfo();
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~TimingFramesLayerInfo();
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size_t target_bitrate_bytes_per_sec = 0;
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std::list<EncodeStartTimeRecord> encode_start_list;
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};
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rtc::CriticalSection lock_;
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EncodedImageCallback* const frame_drop_callback_;
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VideoCodec codec_settings_ RTC_GUARDED_BY(&lock_);
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bool internal_source_ RTC_GUARDED_BY(&lock_);
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uint32_t framerate_fps_ RTC_GUARDED_BY(&lock_);
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// Separate instance for each simulcast stream or spatial layer.
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std::vector<TimingFramesLayerInfo> timing_frames_info_ RTC_GUARDED_BY(&lock_);
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int64_t last_timing_frame_time_ms_ RTC_GUARDED_BY(&lock_);
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size_t incorrect_capture_time_logged_messages_ RTC_GUARDED_BY(&lock_);
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size_t reordered_frames_logged_messages_ RTC_GUARDED_BY(&lock_);
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size_t stalled_encoder_logged_messages_ RTC_GUARDED_BY(&lock_);
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};
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} // namespace webrtc
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#endif // VIDEO_FRAME_ENCODE_TIMER_H_
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