208 lines
7.2 KiB
C++
208 lines
7.2 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 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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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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};
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} // namespace webrtc
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#endif // API_VIDEO_ENCODED_IMAGE_H_
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