1 files changed, 99 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h
new file mode 100644
index 0000000000..f95c3b6c6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -0,0 +1,99 @@
+/*
+ *  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_H264_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <queue>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// Bit masks for NAL (F, NRI, Type) indicators.
+constexpr uint8_t kH264FBit = 0x80;
+constexpr uint8_t kH264NriMask = 0x60;
+constexpr uint8_t kH264TypeMask = 0x1F;
+
+// Bit masks for FU (A and B) headers.
+constexpr uint8_t kH264SBit = 0x80;
+constexpr uint8_t kH264EBit = 0x40;
+constexpr uint8_t kH264RBit = 0x20;
+
+class RtpPacketizerH264 : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded H264 frame.
+  RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
+                    PayloadSizeLimits limits,
+                    H264PacketizationMode packetization_mode);
+
+  ~RtpPacketizerH264() override;
+
+  RtpPacketizerH264(const RtpPacketizerH264&) = delete;
+  RtpPacketizerH264& operator=(const RtpPacketizerH264&) = delete;
+
+  size_t NumPackets() const override;
+
+  // Get the next payload with H264 payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* rtp_packet) override;
+
+ private:
+  // A packet unit (H264 packet), to be put into an RTP packet:
+  // If a NAL unit is too large for an RTP packet, this packet unit will
+  // represent a FU-A packet of a single fragment of the NAL unit.
+  // If a NAL unit is small enough to fit within a single RTP packet, this
+  // packet unit may represent a single NAL unit or a STAP-A packet, of which
+  // there may be multiple in a single RTP packet (if so, aggregated = true).
+  struct PacketUnit {
+    PacketUnit(rtc::ArrayView<const uint8_t> source_fragment,
+               bool first_fragment,
+               bool last_fragment,
+               bool aggregated,
+               uint8_t header)
+        : source_fragment(source_fragment),
+          first_fragment(first_fragment),
+          last_fragment(last_fragment),
+          aggregated(aggregated),
+          header(header) {}
+
+    rtc::ArrayView<const uint8_t> source_fragment;
+    bool first_fragment;
+    bool last_fragment;
+    bool aggregated;
+    uint8_t header;
+  };
+
+  bool GeneratePackets(H264PacketizationMode packetization_mode);
+  bool PacketizeFuA(size_t fragment_index);
+  size_t PacketizeStapA(size_t fragment_index);
+  bool PacketizeSingleNalu(size_t fragment_index);
+
+  void NextAggregatePacket(RtpPacketToSend* rtp_packet);
+  void NextFragmentPacket(RtpPacketToSend* rtp_packet);
+
+  const PayloadSizeLimits limits_;
+  size_t num_packets_left_;
+  std::deque<rtc::ArrayView<const uint8_t>> input_fragments_;
+  std::queue<PacketUnit> packets_;
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_