New class RtxReceiveStream.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2888093002
Cr-Commit-Position: refs/heads/master@{#18212}

webrtc/call/BUILD.gn

@@ -42,6 +42,8 @@ rtc_static_library("call") {
     "rtp_demuxer.cc",
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
+    "rtx_receive_stream.cc",
+    "rtx_receive_stream.h",
   ]
 
   if (!build_with_chromium && is_clang) {
@@ -87,6 +89,7 @@ if (rtc_include_tests) {
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
+      "rtx_receive_stream_unittest.cc",
     ]
     deps = [
       ":call",

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

webrtc/call/rtx_receive_stream.h

@@ -0,0 +1,40 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+
+#include <map>
+
+#include "webrtc/call/rtp_demuxer.h"
+
+namespace webrtc {
+
+class RtxReceiveStream : public RtpPacketSinkInterface {
+ public:
+  RtxReceiveStream(RtpPacketSinkInterface* media_sink,
+                   std::map<int, int> rtx_payload_type_map,
+                   uint32_t media_ssrc);
+
+  // RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+ private:
+  RtpPacketSinkInterface* const media_sink_;
+  // Mapping rtx_payload_type_map_[rtx] = associated.
+  const std::map<int, int> rtx_payload_type_map_;
+  // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
+  // ssrc, and we should delete this.
+  const uint32_t media_ssrc_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_

webrtc/call/rtx_receive_stream_unittest.cc

@@ -0,0 +1,135 @@
+/*
+ *  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 "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::testing::_;
+using ::testing::StrictMock;
+
+class MockRtpPacketSink : public RtpPacketSinkInterface {
+ public:
+  MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&));
+};
+
+constexpr int kMediaPayloadType = 100;
+constexpr int kRtxPayloadType = 98;
+constexpr uint32_t kMediaSSRC = 0x3333333;
+constexpr uint16_t kMediaSeqno = 0x5657;
+
+constexpr uint8_t kRtxPacket[] = {
+    0x80,                    // Version 2.
+    98,                      // Payload type.
+    0x12, 0x34,              // Seqno.
+    0x11, 0x11, 0x11, 0x11,  // Timestamp.
+    0x22, 0x22, 0x22, 0x22,  // SSRC.
+    // RTX header.
+    0x56, 0x57,              // Orig seqno.
+    // Payload.
+    0xee,
+};
+
+constexpr uint8_t kRtxPacketWithCVO[] = {
+    0x90,                    // Version 2, X set.
+    98,                      // Payload type.
+    0x12, 0x34,              // Seqno.
+    0x11, 0x11, 0x11, 0x11,  // Timestamp.
+    0x22, 0x22, 0x22, 0x22,  // SSRC.
+    0xbe, 0xde, 0x00, 0x01,  // Extension header.
+    0x30, 0x01, 0x00, 0x00,  // 90 degree rotation.
+    // RTX header.
+    0x56, 0x57,              // Orig seqno.
+    // Payload.
+    0xee,
+};
+
+std::map<int, int> PayloadTypeMapping() {
+  std::map<int, int> m;
+  m[kRtxPayloadType] = kMediaPayloadType;
+  return m;
+}
+
+template <typename T>
+rtc::ArrayView<T> Truncate(rtc::ArrayView<T> a, size_t drop) {
+  return a.subview(0, a.size() - drop);
+}
+
+}  // namespace
+
+TEST(RtxReceiveStreamTest, RestoresPacketPayload) {
+  StrictMock<MockRtpPacketSink> media_sink;
+  RtxReceiveStream rtx_sink(&media_sink, PayloadTypeMapping(), kMediaSSRC);
+  RtpPacketReceived rtx_packet;
+  EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket)));
+
+  EXPECT_CALL(media_sink, OnRtpPacket(_)).WillOnce(testing::Invoke(
+      [](const RtpPacketReceived& packet) {
+        EXPECT_EQ(packet.SequenceNumber(), kMediaSeqno);
+        EXPECT_EQ(packet.Ssrc(), kMediaSSRC);
+        EXPECT_EQ(packet.PayloadType(), kMediaPayloadType);
+        EXPECT_THAT(packet.payload(), testing::ElementsAre(0xee));
+      }));
+
+  rtx_sink.OnRtpPacket(rtx_packet);
+}
+
+TEST(RtxReceiveStreamTest, IgnoresUnknownPayloadType) {
+  StrictMock<MockRtpPacketSink> media_sink;
+  RtxReceiveStream rtx_sink(&media_sink, std::map<int, int>(), kMediaSSRC);
+  RtpPacketReceived rtx_packet;
+  EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket)));
+  rtx_sink.OnRtpPacket(rtx_packet);
+}
+
+TEST(RtxReceiveStreamTest, IgnoresTruncatedPacket) {
+  StrictMock<MockRtpPacketSink> media_sink;
+  RtxReceiveStream rtx_sink(&media_sink, PayloadTypeMapping(), kMediaSSRC);
+  RtpPacketReceived rtx_packet;
+  EXPECT_TRUE(
+      rtx_packet.Parse(Truncate(rtc::ArrayView<const uint8_t>(kRtxPacket), 2)));
+  rtx_sink.OnRtpPacket(rtx_packet);
+}
+
+TEST(RtxReceiveStreamTest, CopiesRtpHeaderExtensions) {
+  StrictMock<MockRtpPacketSink> media_sink;
+  RtxReceiveStream rtx_sink(&media_sink, PayloadTypeMapping(), kMediaSSRC);
+  RtpHeaderExtensionMap extension_map;
+  extension_map.RegisterByType(3, kRtpExtensionVideoRotation);
+  RtpPacketReceived rtx_packet(&extension_map);
+  EXPECT_TRUE(rtx_packet.Parse(
+      rtc::ArrayView<const uint8_t>(kRtxPacketWithCVO)));
+
+  VideoRotation rotation = kVideoRotation_0;
+  EXPECT_TRUE(rtx_packet.GetExtension<VideoOrientation>(&rotation));
+  EXPECT_EQ(kVideoRotation_90, rotation);
+
+  EXPECT_CALL(media_sink, OnRtpPacket(_)).WillOnce(testing::Invoke(
+      [](const RtpPacketReceived& packet) {
+        EXPECT_EQ(packet.SequenceNumber(), kMediaSeqno);
+        EXPECT_EQ(packet.Ssrc(), kMediaSSRC);
+        EXPECT_EQ(packet.PayloadType(), kMediaPayloadType);
+        EXPECT_THAT(packet.payload(), testing::ElementsAre(0xee));
+        VideoRotation rotation = kVideoRotation_0;
+        EXPECT_TRUE(packet.GetExtension<VideoOrientation>(&rotation));
+        EXPECT_EQ(rotation, kVideoRotation_90);
+      }));
+
+  rtx_sink.OnRtpPacket(rtx_packet);
+}
+
+}  // namespace webrtc