1 files changed, 131 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/video_coding/frame_object.cc b/third_party/libwebrtc/modules/video_coding/frame_object.cc
new file mode 100644
index 0000000000..d226dcd013
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_object.cc
@@ -0,0 +1,131 @@
+/*
+ *  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.
+ */
+
+#include "modules/video_coding/frame_object.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "api/video/encoded_image.h"
+#include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+RtpFrameObject::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)
+    : image_buffer_(image_buffer),
+      first_seq_num_(first_seq_num),
+      last_seq_num_(last_seq_num),
+      last_packet_received_time_(last_packet_received_time),
+      times_nacked_(times_nacked) {
+  rtp_video_header_ = video_header;
+
+  // EncodedFrame members
+  codec_type_ = codec;
+
+  // TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
+  // VCMEncodedFrame members
+  CopyCodecSpecific(&rtp_video_header_);
+  _payloadType = payload_type;
+  SetTimestamp(rtp_timestamp);
+  ntp_time_ms_ = ntp_time_ms;
+  _frameType = rtp_video_header_.frame_type;
+
+  // Setting frame's playout delays to the same values
+  // as of the first packet's.
+  SetPlayoutDelay(rtp_video_header_.playout_delay);
+
+  SetEncodedData(image_buffer_);
+  _encodedWidth = rtp_video_header_.width;
+  _encodedHeight = rtp_video_header_.height;
+
+  // EncodedFrame members
+  SetPacketInfos(std::move(packet_infos));
+
+  rotation_ = rotation;
+  SetColorSpace(color_space);
+  SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
+  content_type_ = content_type;
+  if (timing.flags != VideoSendTiming::kInvalid) {
+    // ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
+    // as this will be dealt with at the time of reporting.
+    timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
+    timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
+    timing_.packetization_finish_ms =
+        ntp_time_ms_ + timing.packetization_finish_delta_ms;
+    timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
+    timing_.network_timestamp_ms =
+        ntp_time_ms_ + timing.network_timestamp_delta_ms;
+    timing_.network2_timestamp_ms =
+        ntp_time_ms_ + timing.network2_timestamp_delta_ms;
+  }
+  timing_.receive_start_ms = first_packet_received_time;
+  timing_.receive_finish_ms = last_packet_received_time;
+  timing_.flags = timing.flags;
+  is_last_spatial_layer = markerBit;
+}
+
+RtpFrameObject::~RtpFrameObject() {
+}
+
+uint16_t RtpFrameObject::first_seq_num() const {
+  return first_seq_num_;
+}
+
+uint16_t RtpFrameObject::last_seq_num() const {
+  return last_seq_num_;
+}
+
+int RtpFrameObject::times_nacked() const {
+  return times_nacked_;
+}
+
+VideoFrameType RtpFrameObject::frame_type() const {
+  return rtp_video_header_.frame_type;
+}
+
+VideoCodecType RtpFrameObject::codec_type() const {
+  return codec_type_;
+}
+
+int64_t RtpFrameObject::ReceivedTime() const {
+  return last_packet_received_time_;
+}
+
+int64_t RtpFrameObject::RenderTime() const {
+  return _renderTimeMs;
+}
+
+bool RtpFrameObject::delayed_by_retransmission() const {
+  return times_nacked() > 0;
+}
+
+const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
+  return rtp_video_header_;
+}
+
+}  // namespace webrtc