Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 68 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/video_coding/frame_object.h b/third_party/libwebrtc/modules/video_coding/frame_object.h
new file mode 100644
index 0000000000..c6f069f241
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_object.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_VIDEO_CODING_FRAME_OBJECT_H_
+#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_frame.h"
+
+namespace webrtc {
+
+class RtpFrameObject : public EncodedFrame {
+ public:
+  RtpFrameObject(uint16_t first_seq_num,
+                 uint16_t last_seq_num,
+                 bool markerBit,
+                 int times_nacked,
+                 int64_t first_packet_received_time,
+                 int64_t last_packet_received_time,
+                 uint32_t rtp_timestamp,
+                 int64_t ntp_time_ms,
+                 const VideoSendTiming& timing,
+                 uint8_t payload_type,
+                 VideoCodecType codec,
+                 VideoRotation rotation,
+                 VideoContentType content_type,
+                 const RTPVideoHeader& video_header,
+                 const absl::optional<webrtc::ColorSpace>& color_space,
+                 RtpPacketInfos packet_infos,
+                 rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
+
+  ~RtpFrameObject() override;
+  uint16_t first_seq_num() const;
+  uint16_t last_seq_num() const;
+  int times_nacked() const;
+  VideoFrameType frame_type() const;
+  VideoCodecType codec_type() const;
+  int64_t ReceivedTime() const override;
+  int64_t RenderTime() const override;
+  bool delayed_by_retransmission() const override;
+  const RTPVideoHeader& GetRtpVideoHeader() const;
+
+  uint8_t* mutable_data() { return image_buffer_->data(); }
+
+ private:
+  // Reference for mutable access.
+  rtc::scoped_refptr<EncodedImageBuffer> image_buffer_;
+  RTPVideoHeader rtp_video_header_;
+  VideoCodecType codec_type_;
+  uint16_t first_seq_num_;
+  uint16_t last_seq_num_;
+  int64_t last_packet_received_time_;
+
+  // Equal to times nacked of the packet with the highet times nacked
+  // belonging to this frame.
+  int times_nacked_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_OBJECT_H_