Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 111 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h
new file mode 100644
index 0000000000..07deb0f9bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2014 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.
+ *
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace rtcp {
+// Class for building RTCP packets.
+//
+//  Example:
+//  ReportBlock report_block;
+//  report_block.SetMediaSsrc(234);
+//  report_block.SetFractionLost(10);
+//
+//  ReceiverReport rr;
+//  rr.SetSenderSsrc(123);
+//  rr.AddReportBlock(report_block);
+//
+//  Fir fir;
+//  fir.SetSenderSsrc(123);
+//  fir.AddRequestTo(234, 56);
+//
+//  size_t length = 0;                     // Builds an intra frame request
+//  uint8_t packet[kPacketSize];           // with sequence number 56.
+//  fir.Build(packet, &length, kPacketSize);
+//
+//  rtc::Buffer packet = fir.Build();      // Returns a RawPacket holding
+//                                         // the built rtcp packet.
+//
+//  CompoundPacket compound;               // Builds a compound RTCP packet with
+//  compound.Append(&rr);                  // a receiver report, report block
+//  compound.Append(&fir);                 // and fir message.
+//  rtc::Buffer packet = compound.Build();
+
+class RtcpPacket {
+ public:
+  // Callback used to signal that an RTCP packet is ready. Note that this may
+  // not contain all data in this RtcpPacket; if a packet cannot fit in
+  // max_length bytes, it will be fragmented and multiple calls to this
+  // callback will be made.
+  using PacketReadyCallback =
+      rtc::FunctionView<void(rtc::ArrayView<const uint8_t> packet)>;
+
+  virtual ~RtcpPacket() = default;
+
+  void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+
+  // Convenience method mostly used for test. Creates packet without
+  // fragmentation using BlockLength() to allocate big enough buffer.
+  rtc::Buffer Build() const;
+
+  // Returns true if call to Create succeeded.
+  bool Build(size_t max_length, PacketReadyCallback callback) const;
+
+  // Size of this packet in bytes (including headers).
+  virtual size_t BlockLength() const = 0;
+
+  // Creates packet in the given buffer at the given position.
+  // Calls PacketReadyCallback::OnPacketReady if remaining buffer is too small
+  // and assume buffer can be reused after OnPacketReady returns.
+  virtual bool Create(uint8_t* packet,
+                      size_t* index,
+                      size_t max_length,
+                      PacketReadyCallback callback) const = 0;
+
+ protected:
+  // Size of the rtcp common header.
+  static constexpr size_t kHeaderLength = 4;
+  RtcpPacket() {}
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           bool padding,  // True if there are padding bytes.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  bool OnBufferFull(uint8_t* packet,
+                    size_t* index,
+                    PacketReadyCallback callback) const;
+  // Size of the rtcp packet as written in header.
+  size_t HeaderLength() const;
+
+ private:
+  uint32_t sender_ssrc_ = 0;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_