New class RtxReceiveStream.

webrtc/call/rtx_receive_stream.cc

Index: webrtc/call/rtx_receive_stream.cc
diff --git a/webrtc/call/rtx_receive_stream.cc b/webrtc/call/rtx_receive_stream.cc
new file mode 100644
index 0000000000000000000000000000000000000000..286f867faa8985f6bf5013098ee4041c9f23e8ea
--- /dev/null
+++ b/webrtc/call/rtx_receive_stream.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 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 <utility>
+
+#include "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+RtxReceiveStream::RtxReceiveStream(
+    RtpPacketSinkInterface* media_sink,
+    std::map<int, int> rtx_payload_type_map,
+    uint32_t media_ssrc)
+    : media_sink_(media_sink),
+      rtx_payload_type_map_(std::move(rtx_payload_type_map)),
+      media_ssrc_(media_ssrc) {}
+
+void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
+  rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
+
+  if (payload.size() < kRtxHeaderSize) {
+    return;
+  }
+
+  auto it = rtx_payload_type_map_.find(rtx_packet.PayloadType());
+  if (it == rtx_payload_type_map_.end()) {
+    return;
+  }
+  RtpPacketReceived media_packet;
+  media_packet.CopyHeaderFrom(rtx_packet);
+
+  media_packet.SetSsrc(media_ssrc_);
+  media_packet.SetSequenceNumber((payload[0] << 8) + payload[1]);
+  media_packet.SetPayloadType(it->second);
+
+  // Skip the RTX header.
+  rtc::ArrayView<const uint8_t> rtx_payload =
+      payload.subview(kRtxHeaderSize);
+
+  uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
+  RTC_DCHECK(media_payload != nullptr);
+
+  memcpy(media_payload, rtx_payload.data(), rtx_payload.size());
+
+  media_sink_->OnRtpPacket(media_packet);
+}
+
+}  // namespace webrtc