16a8792e0a
Before this, an empty list of CSRCs was always provided up to encoded insertable streams transforms for remote video tracks, regardless of the actual CSRCs on received frames. Audio already works correctly. Bug: chromium:1411614 Change-Id: I51ab4dc5e67a1a35893fefff16c1f057e9047e6b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291539 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#39240}
136 lines
4.0 KiB
C++
136 lines
4.0 KiB
C++
/*
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* Copyright (c) 2016 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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#include "modules/video_coding/frame_object.h"
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#include <string.h>
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#include <utility>
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#include "api/video/encoded_image.h"
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#include "api/video/video_timing.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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RtpFrameObject::RtpFrameObject(
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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bool markerBit,
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int times_nacked,
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int64_t first_packet_received_time,
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int64_t last_packet_received_time,
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uint32_t rtp_timestamp,
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int64_t ntp_time_ms,
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const VideoSendTiming& timing,
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uint8_t payload_type,
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VideoCodecType codec,
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VideoRotation rotation,
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
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: image_buffer_(image_buffer),
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first_seq_num_(first_seq_num),
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last_seq_num_(last_seq_num),
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last_packet_received_time_(last_packet_received_time),
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times_nacked_(times_nacked) {
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rtp_video_header_ = video_header;
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// EncodedFrame members
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codec_type_ = codec;
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// TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
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// VCMEncodedFrame members
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CopyCodecSpecific(&rtp_video_header_);
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_payloadType = payload_type;
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SetTimestamp(rtp_timestamp);
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ntp_time_ms_ = ntp_time_ms;
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_frameType = rtp_video_header_.frame_type;
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// Setting frame's playout delays to the same values
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// as of the first packet's.
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SetPlayoutDelay(rtp_video_header_.playout_delay);
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SetEncodedData(image_buffer_);
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_encodedWidth = rtp_video_header_.width;
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_encodedHeight = rtp_video_header_.height;
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if (packet_infos.begin() != packet_infos.end()) {
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csrcs_ = packet_infos.begin()->csrcs();
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}
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// EncodedFrame members
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SetPacketInfos(std::move(packet_infos));
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rotation_ = rotation;
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SetColorSpace(color_space);
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SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
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content_type_ = content_type;
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if (timing.flags != VideoSendTiming::kInvalid) {
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// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
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// as this will be dealt with at the time of reporting.
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timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
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timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
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timing_.packetization_finish_ms =
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ntp_time_ms_ + timing.packetization_finish_delta_ms;
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timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
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timing_.network_timestamp_ms =
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ntp_time_ms_ + timing.network_timestamp_delta_ms;
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timing_.network2_timestamp_ms =
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ntp_time_ms_ + timing.network2_timestamp_delta_ms;
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}
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timing_.receive_start_ms = first_packet_received_time;
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timing_.receive_finish_ms = last_packet_received_time;
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timing_.flags = timing.flags;
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is_last_spatial_layer = markerBit;
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}
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RtpFrameObject::~RtpFrameObject() {
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}
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uint16_t RtpFrameObject::first_seq_num() const {
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return first_seq_num_;
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}
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uint16_t RtpFrameObject::last_seq_num() const {
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return last_seq_num_;
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}
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int RtpFrameObject::times_nacked() const {
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return times_nacked_;
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}
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VideoFrameType RtpFrameObject::frame_type() const {
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return rtp_video_header_.frame_type;
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}
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VideoCodecType RtpFrameObject::codec_type() const {
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return codec_type_;
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}
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int64_t RtpFrameObject::ReceivedTime() const {
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return last_packet_received_time_;
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}
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int64_t RtpFrameObject::RenderTime() const {
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return _renderTimeMs;
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}
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bool RtpFrameObject::delayed_by_retransmission() const {
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return times_nacked() > 0;
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}
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const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
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return rtp_video_header_;
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}
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} // namespace webrtc
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