Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 69 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/video_coding/packet.cc b/third_party/libwebrtc/modules/video_coding/packet.cc
new file mode 100644
index 0000000000..f1bac4a305
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 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/packet.h"
+
+#include "api/rtp_headers.h"
+
+namespace webrtc {
+
+VCMPacket::VCMPacket()
+    : payloadType(0),
+      timestamp(0),
+      ntp_time_ms_(0),
+      seqNum(0),
+      dataPtr(NULL),
+      sizeBytes(0),
+      markerBit(false),
+      timesNacked(-1),
+      completeNALU(kNaluUnset),
+      insertStartCode(false),
+      video_header() {
+  video_header.playout_delay = {-1, -1};
+}
+
+VCMPacket::VCMPacket(const uint8_t* ptr,
+                     size_t size,
+                     const RTPHeader& rtp_header,
+                     const RTPVideoHeader& videoHeader,
+                     int64_t ntp_time_ms,
+                     Timestamp receive_time)
+    : payloadType(rtp_header.payloadType),
+      timestamp(rtp_header.timestamp),
+      ntp_time_ms_(ntp_time_ms),
+      seqNum(rtp_header.sequenceNumber),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(rtp_header.markerBit),
+      timesNacked(-1),
+      completeNALU(kNaluIncomplete),
+      insertStartCode(videoHeader.codec == kVideoCodecH264 &&
+                      videoHeader.is_first_packet_in_frame),
+      video_header(videoHeader),
+      packet_info(rtp_header, receive_time) {
+  if (is_first_packet_in_frame() && markerBit) {
+    completeNALU = kNaluComplete;
+  } else if (is_first_packet_in_frame()) {
+    completeNALU = kNaluStart;
+  } else if (markerBit) {
+    completeNALU = kNaluEnd;
+  } else {
+    completeNALU = kNaluIncomplete;
+  }
+
+  // Playout decisions are made entirely based on first packet in a frame.
+  if (!is_first_packet_in_frame()) {
+    video_header.playout_delay = {-1, -1};
+  }
+}
+
+VCMPacket::~VCMPacket() = default;
+
+}  // namespace webrtc