Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 226 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/video/encoded_image.h b/third_party/libwebrtc/api/video/encoded_image.h
new file mode 100644
index 0000000000..dae790c46c
--- /dev/null
+++ b/third_party/libwebrtc/api/video/encoded_image.h
@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef API_VIDEO_ENCODED_IMAGE_H_
+#define API_VIDEO_ENCODED_IMAGE_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/rtp_packet_infos.h"
+#include "api/scoped_refptr.h"
+#include "api/video/color_space.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_frame_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Abstract interface for buffer storage. Intended to support buffers owned by
+// external encoders with special release requirements, e.g, java encoders with
+// releaseOutputBuffer.
+class EncodedImageBufferInterface : public rtc::RefCountInterface {
+ public:
+  virtual const uint8_t* data() const = 0;
+  // TODO(bugs.webrtc.org/9378): Make interface essentially read-only, delete
+  // this non-const data method.
+  virtual uint8_t* data() = 0;
+  virtual size_t size() const = 0;
+};
+
+// Basic implementation of EncodedImageBufferInterface.
+class RTC_EXPORT EncodedImageBuffer : public EncodedImageBufferInterface {
+ public:
+  static rtc::scoped_refptr<EncodedImageBuffer> Create() { return Create(0); }
+  static rtc::scoped_refptr<EncodedImageBuffer> Create(size_t size);
+  static rtc::scoped_refptr<EncodedImageBuffer> Create(const uint8_t* data,
+                                                       size_t size);
+
+  const uint8_t* data() const override;
+  uint8_t* data() override;
+  size_t size() const override;
+  void Realloc(size_t t);
+
+ protected:
+  explicit EncodedImageBuffer(size_t size);
+  EncodedImageBuffer(const uint8_t* data, size_t size);
+  ~EncodedImageBuffer();
+
+  size_t size_;
+  uint8_t* buffer_;
+};
+
+// TODO(bug.webrtc.org/9378): This is a legacy api class, which is slowly being
+// cleaned up. Direct use of its members is strongly discouraged.
+class RTC_EXPORT EncodedImage {
+ public:
+  EncodedImage();
+  EncodedImage(EncodedImage&&);
+  EncodedImage(const EncodedImage&);
+
+  ~EncodedImage();
+
+  EncodedImage& operator=(EncodedImage&&);
+  EncodedImage& operator=(const EncodedImage&);
+
+  // TODO(bugs.webrtc.org/9378): Change style to timestamp(), set_timestamp(),
+  // for consistency with the VideoFrame class. Set frame timestamp (90kHz).
+  void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
+
+  // Get frame timestamp (90kHz).
+  uint32_t Timestamp() const { return timestamp_rtp_; }
+
+  void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
+
+  int64_t NtpTimeMs() const { return ntp_time_ms_; }
+
+  absl::optional<int> SpatialIndex() const { return spatial_index_; }
+  void SetSpatialIndex(absl::optional<int> spatial_index) {
+    RTC_DCHECK_GE(spatial_index.value_or(0), 0);
+    RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
+    spatial_index_ = spatial_index;
+  }
+
+  absl::optional<int> TemporalIndex() const { return temporal_index_; }
+  void SetTemporalIndex(absl::optional<int> temporal_index) {
+    RTC_DCHECK_GE(temporal_index_.value_or(0), 0);
+    RTC_DCHECK_LT(temporal_index_.value_or(0), kMaxTemporalStreams);
+    temporal_index_ = temporal_index;
+  }
+
+  // These methods can be used to set/get size of subframe with spatial index
+  // `spatial_index` on encoded frames that consist of multiple spatial layers.
+  absl::optional<size_t> SpatialLayerFrameSize(int spatial_index) const;
+  void SetSpatialLayerFrameSize(int spatial_index, size_t size_bytes);
+
+  const webrtc::ColorSpace* ColorSpace() const {
+    return color_space_ ? &*color_space_ : nullptr;
+  }
+  void SetColorSpace(const absl::optional<webrtc::ColorSpace>& color_space) {
+    color_space_ = color_space;
+  }
+
+  // These methods along with the private member video_frame_tracking_id_ are
+  // meant for media quality testing purpose only.
+  absl::optional<uint16_t> VideoFrameTrackingId() const {
+    return video_frame_tracking_id_;
+  }
+  void SetVideoFrameTrackingId(absl::optional<uint16_t> tracking_id) {
+    video_frame_tracking_id_ = tracking_id;
+  }
+
+  const RtpPacketInfos& PacketInfos() const { return packet_infos_; }
+  void SetPacketInfos(RtpPacketInfos packet_infos) {
+    packet_infos_ = std::move(packet_infos);
+  }
+
+  bool RetransmissionAllowed() const { return retransmission_allowed_; }
+  void SetRetransmissionAllowed(bool retransmission_allowed) {
+    retransmission_allowed_ = retransmission_allowed;
+  }
+
+  size_t size() const { return size_; }
+  void set_size(size_t new_size) {
+    // Allow set_size(0) even if we have no buffer.
+    RTC_DCHECK_LE(new_size, new_size == 0 ? 0 : capacity());
+    size_ = new_size;
+  }
+
+  void SetEncodedData(
+      rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data) {
+    encoded_data_ = encoded_data;
+    size_ = encoded_data->size();
+  }
+
+  void ClearEncodedData() {
+    encoded_data_ = nullptr;
+    size_ = 0;
+  }
+
+  rtc::scoped_refptr<EncodedImageBufferInterface> GetEncodedData() const {
+    return encoded_data_;
+  }
+
+  const uint8_t* data() const {
+    return encoded_data_ ? encoded_data_->data() : nullptr;
+  }
+
+  // Returns whether the encoded image can be considered to be of target
+  // quality.
+  bool IsAtTargetQuality() const { return at_target_quality_; }
+
+  // Sets that the encoded image can be considered to be of target quality to
+  // true or false.
+  void SetAtTargetQuality(bool at_target_quality) {
+    at_target_quality_ = at_target_quality;
+  }
+
+  uint32_t _encodedWidth = 0;
+  uint32_t _encodedHeight = 0;
+  // NTP time of the capture time in local timebase in milliseconds.
+  // TODO(minyue): make this member private.
+  int64_t ntp_time_ms_ = 0;
+  int64_t capture_time_ms_ = 0;
+  VideoFrameType _frameType = VideoFrameType::kVideoFrameDelta;
+  VideoRotation rotation_ = kVideoRotation_0;
+  VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
+  int qp_ = -1;  // Quantizer value.
+
+  // When an application indicates non-zero values here, it is taken as an
+  // indication that all future frames will be constrained with those limits
+  // until the application indicates a change again.
+  VideoPlayoutDelay playout_delay_;
+
+  struct Timing {
+    uint8_t flags = VideoSendTiming::kInvalid;
+    int64_t encode_start_ms = 0;
+    int64_t encode_finish_ms = 0;
+    int64_t packetization_finish_ms = 0;
+    int64_t pacer_exit_ms = 0;
+    int64_t network_timestamp_ms = 0;
+    int64_t network2_timestamp_ms = 0;
+    int64_t receive_start_ms = 0;
+    int64_t receive_finish_ms = 0;
+  } timing_;
+
+ private:
+  size_t capacity() const { return encoded_data_ ? encoded_data_->size() : 0; }
+
+  rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
+  size_t size_ = 0;  // Size of encoded frame data.
+  uint32_t timestamp_rtp_ = 0;
+  absl::optional<int> spatial_index_;
+  absl::optional<int> temporal_index_;
+  std::map<int, size_t> spatial_layer_frame_size_bytes_;
+  absl::optional<webrtc::ColorSpace> color_space_;
+  // This field is meant for media quality testing purpose only. When enabled it
+  // carries the webrtc::VideoFrame id field from the sender to the receiver.
+  absl::optional<uint16_t> video_frame_tracking_id_;
+  // Information about packets used to assemble this video frame. This is needed
+  // by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
+  // MediaStreamTrack, in order to implement getContributingSources(). See:
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
+  RtpPacketInfos packet_infos_;
+  bool retransmission_allowed_ = true;
+  // True if the encoded image can be considered to be of target quality.
+  bool at_target_quality_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_ENCODED_IMAGE_H_