1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h
new file mode 100644
index 0000000000..95442f795c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2023 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_RTP_PACKETIZER_H265_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_
+
+#include <deque>
+#include <queue>
+#include <string>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class RtpPacketizerH265 : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded H.265 frame.
+  // For H265 we only support tx-mode SRST.
+  RtpPacketizerH265(rtc::ArrayView<const uint8_t> payload,
+                    PayloadSizeLimits limits);
+
+  RtpPacketizerH265(const RtpPacketizerH265&) = delete;
+  RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete;
+
+  ~RtpPacketizerH265() override;
+
+  size_t NumPackets() const override;
+
+  // Get the next payload with H.265 payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success or false if there was no payload to packetize.
+  bool NextPacket(RtpPacketToSend* rtp_packet) override;
+
+ private:
+  struct PacketUnit {
+    rtc::ArrayView<const uint8_t> source_fragment;
+    bool first_fragment = false;
+    bool last_fragment = false;
+    bool aggregated = false;
+    uint16_t header = 0;
+  };
+  std::deque<rtc::ArrayView<const uint8_t>> input_fragments_;
+  std::queue<PacketUnit> packets_;
+
+  bool GeneratePackets();
+  bool PacketizeFu(size_t fragment_index);
+  int PacketizeAp(size_t fragment_index);
+
+  void NextAggregatePacket(RtpPacketToSend* rtp_packet);
+  void NextFragmentPacket(RtpPacketToSend* rtp_packet);
+
+  const PayloadSizeLimits limits_;
+  size_t num_packets_left_ = 0;
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_