| /* |
| * 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" |
| #include "rtc_base/critical_section.h" |
| |
| namespace webrtc { |
| namespace video_coding { |
| 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) |
| : 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_); |
| _completeFrame = true; |
| _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(std::move(image_buffer)); |
| _encodedWidth = rtp_video_header_.width; |
| _encodedHeight = rtp_video_header_.height; |
| |
| // EncodedFrame members |
| SetPacketInfos(std::move(packet_infos)); |
| |
| rotation_ = rotation; |
| SetColorSpace(color_space); |
| 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 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, |
| const absl::optional<RtpGenericFrameDescriptor>& /*generic_descriptor*/, |
| RtpPacketInfos packet_infos, |
| rtc::scoped_refptr<EncodedImageBuffer> image_buffer) |
| : RtpFrameObject(first_seq_num, |
| last_seq_num, |
| markerBit, |
| times_nacked, |
| first_packet_received_time, |
| last_packet_received_time, |
| rtp_timestamp, |
| ntp_time_ms, |
| timing, |
| payload_type, |
| codec, |
| rotation, |
| content_type, |
| video_header, |
| color_space, |
| std::move(packet_infos), |
| std::move(image_buffer)) {} |
| |
| 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_; |
| } |
| |
| const FrameMarking& RtpFrameObject::GetFrameMarking() const { |
| return rtp_video_header_.frame_marking; |
| } |
| |
| } // namespace video_coding |
| } // namespace webrtc |