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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/video_coding/frame_object.cc
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/video_coding/frame_object.cc')
-rw-r--r--third_party/libwebrtc/modules/video_coding/frame_object.cc131
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
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+++ 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