New class RtxReceiveStream.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2888093002
Cr-Commit-Position: refs/heads/master@{#18212}
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 0d87b2f..7728104 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -42,6 +42,8 @@
"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 @@
"bitrate_estimator_tests.cc",
"call_unittest.cc",
"flexfec_receive_stream_unittest.cc",
+ "rtx_receive_stream_unittest.cc",
]
deps = [
":call",
diff --git a/webrtc/call/rtx_receive_stream.cc b/webrtc/call/rtx_receive_stream.cc
new file mode 100644
index 0000000..286f867
--- /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
diff --git a/webrtc/call/rtx_receive_stream.h b/webrtc/call/rtx_receive_stream.h
new file mode 100644
index 0000000..6e02851
--- /dev/null
+++ b/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_
diff --git a/webrtc/call/rtx_receive_stream_unittest.cc b/webrtc/call/rtx_receive_stream_unittest.cc
new file mode 100644
index 0000000..6bb067b
--- /dev/null
+++ b/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