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The frame cadence adapter previously resulted in unconditional frame repeating at max FPS. Change this to slow down to an idle rate (1 Hz) when quality convergence in all configured spatial layers has been achieved. go/rtc-0hz-present Bug: chromium:1255737 Change-Id: Ifa593dbf8a61aa29da20ac250da332734ae82791 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241421 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35547}
219 lines
7.6 KiB
C++
219 lines
7.6 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 API_VIDEO_ENCODED_IMAGE_H_
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#define API_VIDEO_ENCODED_IMAGE_H_
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#include <stdint.h>
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#include <map>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/rtp_packet_infos.h"
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#include "api/scoped_refptr.h"
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#include "api/video/color_space.h"
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#include "api/video/video_codec_constants.h"
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#include "api/video/video_content_type.h"
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#include "api/video/video_frame_type.h"
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#include "api/video/video_rotation.h"
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#include "api/video/video_timing.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// Abstract interface for buffer storage. Intended to support buffers owned by
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// external encoders with special release requirements, e.g, java encoders with
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// releaseOutputBuffer.
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class EncodedImageBufferInterface : public rtc::RefCountInterface {
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public:
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virtual const uint8_t* data() const = 0;
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// TODO(bugs.webrtc.org/9378): Make interface essentially read-only, delete
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// this non-const data method.
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virtual uint8_t* data() = 0;
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virtual size_t size() const = 0;
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};
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// Basic implementation of EncodedImageBufferInterface.
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class RTC_EXPORT EncodedImageBuffer : public EncodedImageBufferInterface {
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public:
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static rtc::scoped_refptr<EncodedImageBuffer> Create() { return Create(0); }
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static rtc::scoped_refptr<EncodedImageBuffer> Create(size_t size);
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static rtc::scoped_refptr<EncodedImageBuffer> Create(const uint8_t* data,
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size_t size);
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const uint8_t* data() const override;
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uint8_t* data() override;
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size_t size() const override;
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void Realloc(size_t t);
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protected:
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explicit EncodedImageBuffer(size_t size);
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EncodedImageBuffer(const uint8_t* data, size_t size);
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~EncodedImageBuffer();
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size_t size_;
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uint8_t* buffer_;
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};
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// TODO(bug.webrtc.org/9378): This is a legacy api class, which is slowly being
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// cleaned up. Direct use of its members is strongly discouraged.
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class RTC_EXPORT EncodedImage {
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public:
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EncodedImage();
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EncodedImage(EncodedImage&&);
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EncodedImage(const EncodedImage&);
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~EncodedImage();
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EncodedImage& operator=(EncodedImage&&);
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EncodedImage& operator=(const EncodedImage&);
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// TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
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// with the VideoFrame class.
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// Set frame timestamp (90kHz).
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void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
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// Get frame timestamp (90kHz).
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uint32_t Timestamp() const { return timestamp_rtp_; }
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void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
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int64_t NtpTimeMs() const { return ntp_time_ms_; }
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absl::optional<int> SpatialIndex() const { return spatial_index_; }
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void SetSpatialIndex(absl::optional<int> spatial_index) {
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RTC_DCHECK_GE(spatial_index.value_or(0), 0);
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RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
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spatial_index_ = spatial_index;
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}
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// These methods can be used to set/get size of subframe with spatial index
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// `spatial_index` on encoded frames that consist of multiple spatial layers.
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absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
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void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
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const webrtc::ColorSpace* ColorSpace() const {
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return color_space_ ? &*color_space_ : nullptr;
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}
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void SetColorSpace(const absl::optional<webrtc::ColorSpace>& color_space) {
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color_space_ = color_space;
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}
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// These methods along with the private member video_frame_tracking_id_ are
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// meant for media quality testing purpose only.
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absl::optional<uint16_t> VideoFrameTrackingId() const {
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return video_frame_tracking_id_;
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}
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void SetVideoFrameTrackingId(absl::optional<uint16_t> tracking_id) {
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video_frame_tracking_id_ = tracking_id;
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}
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const RtpPacketInfos& PacketInfos() const { return packet_infos_; }
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void SetPacketInfos(RtpPacketInfos packet_infos) {
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packet_infos_ = std::move(packet_infos);
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}
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bool RetransmissionAllowed() const { return retransmission_allowed_; }
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void SetRetransmissionAllowed(bool retransmission_allowed) {
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retransmission_allowed_ = retransmission_allowed;
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}
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size_t size() const { return size_; }
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void set_size(size_t new_size) {
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// Allow set_size(0) even if we have no buffer.
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RTC_DCHECK_LE(new_size, new_size == 0 ? 0 : capacity());
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size_ = new_size;
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}
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void SetEncodedData(
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rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data) {
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encoded_data_ = encoded_data;
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size_ = encoded_data->size();
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}
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void ClearEncodedData() {
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encoded_data_ = nullptr;
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size_ = 0;
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}
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rtc::scoped_refptr<EncodedImageBufferInterface> GetEncodedData() const {
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return encoded_data_;
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}
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const uint8_t* data() const {
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return encoded_data_ ? encoded_data_->data() : nullptr;
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}
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// Returns whether the encoded image can be considered to be of target
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// quality.
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bool IsAtTargetQuality() const { return at_target_quality_; }
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// Sets that the encoded image can be considered to be of target quality to
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// true or false.
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void SetAtTargetQuality(bool at_target_quality) {
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at_target_quality_ = at_target_quality;
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}
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uint32_t _encodedWidth = 0;
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uint32_t _encodedHeight = 0;
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// NTP time of the capture time in local timebase in milliseconds.
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// TODO(minyue): make this member private.
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int64_t ntp_time_ms_ = 0;
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int64_t capture_time_ms_ = 0;
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VideoFrameType _frameType = VideoFrameType::kVideoFrameDelta;
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VideoRotation rotation_ = kVideoRotation_0;
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VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
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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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VideoPlayoutDelay playout_delay_;
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struct Timing {
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uint8_t flags = VideoSendTiming::kInvalid;
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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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private:
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size_t capacity() const { return encoded_data_ ? encoded_data_->size() : 0; }
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rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
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size_t size_ = 0; // Size of encoded frame data.
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uint32_t timestamp_rtp_ = 0;
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absl::optional<int> spatial_index_;
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std::map<int, size_t> spatial_layer_frame_size_bytes_;
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absl::optional<webrtc::ColorSpace> color_space_;
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// This field is meant for media quality testing purpose only. When enabled it
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// carries the webrtc::VideoFrame id field from the sender to the receiver.
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absl::optional<uint16_t> video_frame_tracking_id_;
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// Information about packets used to assemble this video frame. This is needed
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// by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
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// MediaStreamTrack, in order to implement getContributingSources(). See:
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
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RtpPacketInfos packet_infos_;
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bool retransmission_allowed_ = true;
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// True if the encoded image can be considered to be of target quality.
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bool at_target_quality_ = false;
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};
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} // namespace webrtc
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#endif // API_VIDEO_ENCODED_IMAGE_H_
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