1 files changed, 61 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h
new file mode 100644
index 0000000000..19abd3feb2
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+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -0,0 +1,61 @@
+/*
+ *  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_RTP_FORMAT_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+class RtpPacketToSend;
+
+class RtpPacketizer {
+ public:
+  struct PayloadSizeLimits {
+    int max_payload_len = 1200;
+    int first_packet_reduction_len = 0;
+    int last_packet_reduction_len = 0;
+    // Reduction len for packet that is first & last at the same time.
+    int single_packet_reduction_len = 0;
+  };
+
+  // If type is not set, returns a raw packetizer.
+  static std::unique_ptr<RtpPacketizer> Create(
+      absl::optional<VideoCodecType> type,
+      rtc::ArrayView<const uint8_t> payload,
+      PayloadSizeLimits limits,
+      // Codec-specific details.
+      const RTPVideoHeader& rtp_video_header);
+
+  virtual ~RtpPacketizer() = default;
+
+  // Returns number of remaining packets to produce by the packetizer.
+  virtual size_t NumPackets() const = 0;
+
+  // Get the next payload with payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  virtual bool NextPacket(RtpPacketToSend* packet) = 0;
+
+  // Split payload_len into sum of integers with respect to `limits`.
+  // Returns empty vector on failure.
+  static std::vector<int> SplitAboutEqually(int payload_len,
+                                            const PayloadSizeLimits& limits);
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_