Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ )
Reason for revert:
The cl this change depended on has now been successfully relanded.
Original issue's description:
> Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ )
>
> Reason for revert:
> This has to be reverted to enable reverting cl https://codereview.webrtc.org/3006063002/, which seems to have broken ulpfec.
>
> Original issue's description:
> > Delete Rtx-related methods from RTPPayloadRegistry.
> >
> > Delete methods IsRtx, IsEncapsulated and RestoreOriginalPacket.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/3006993002
> > Cr-Commit-Position: refs/heads/master@{#19739}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/5b4b52264132eefba10bc6880871715f9692da90
>
> TBR=stefan@webrtc.org,danilchap@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3011093002
> Cr-Commit-Position: refs/heads/master@{#19742}
> Committed: https://chromium.googlesource.com/external/webrtc/+/a64685325c2f8f51873b67ae8a91f94ffb19d70b
TBR=stefan@webrtc.org,danilchap@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3012253002
Cr-Original-Commit-Position: refs/heads/master@{#19821}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: f54573bd3b9774db6782af8de0f22242598cb974
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 82bd16d..811dca6 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -368,6 +368,7 @@
"../../api:array_view",
"../../api:libjingle_peerconnection_api",
"../../api:transport_api",
+ "../../call:rtp_receiver",
"../../common_video:common_video",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
diff --git a/modules/rtp_rtcp/include/rtp_payload_registry.h b/modules/rtp_rtcp/include/rtp_payload_registry.h
index a84e2d3..4eda28d 100644
--- a/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -53,20 +53,8 @@
void SetRtxPayloadType(int payload_type, int associated_payload_type);
- bool IsRtx(const RTPHeader& header) const;
-
- bool RestoreOriginalPacket(uint8_t* restored_packet,
- const uint8_t* packet,
- size_t* packet_length,
- uint32_t original_ssrc,
- const RTPHeader& header);
-
bool IsRed(const RTPHeader& header) const;
- // Returns true if the media of this RTP packet is encapsulated within an
- // extra header, such as RTX or RED.
- bool IsEncapsulated(const RTPHeader& header) const;
-
bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
int GetPayloadTypeFrequency(uint8_t payload_type) const;
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 32c9d5b..1a68726 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -15,13 +15,13 @@
#include <set>
#include "webrtc/api/call/transport.h"
+#include "webrtc/call/rtp_stream_receiver_controller.h"
+#include "webrtc/call/rtx_receive_stream.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/rtc_base/rate_limiter.h"
#include "webrtc/test/gtest.h"
@@ -29,6 +29,7 @@
const int kVideoNackListSize = 30;
const uint32_t kTestSsrc = 3456;
+const uint32_t kTestRtxSsrc = kTestSsrc + 1;
const uint16_t kTestSequenceNumber = 2345;
const uint32_t kTestNumberOfPackets = 1350;
const int kTestNumberOfRtxPackets = 149;
@@ -37,35 +38,19 @@
const int kRtxPayloadType = 98;
const int64_t kMaxRttMs = 1000;
-class VerifyingRtxReceiver : public RtpData {
+class VerifyingMediaStream : public RtpPacketSinkInterface {
public:
- VerifyingRtxReceiver() {}
+ VerifyingMediaStream() {}
- int32_t OnReceivedPayloadData(
- const uint8_t* data,
- size_t size,
- const webrtc::WebRtcRTPHeader* rtp_header) override {
+ void OnRtpPacket(const RtpPacketReceived& packet) override {
if (!sequence_numbers_.empty())
- EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
- sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
- return 0;
+ EXPECT_EQ(kTestSsrc, packet.Ssrc());
+
+ sequence_numbers_.push_back(packet.SequenceNumber());
}
std::list<uint16_t> sequence_numbers_;
};
-class TestRtpFeedback : public NullRtpFeedback {
- public:
- explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
- virtual ~TestRtpFeedback() {}
-
- void OnIncomingSSRCChanged(uint32_t ssrc) override {
- rtp_rtcp_->SetRemoteSSRC(ssrc);
- }
-
- private:
- RtpRtcp* rtp_rtcp_;
-};
-
class RtxLoopBackTransport : public webrtc::Transport {
public:
explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
@@ -75,16 +60,10 @@
consecutive_drop_end_(0),
rtx_ssrc_(rtx_ssrc),
count_rtx_ssrc_(0),
- rtp_payload_registry_(NULL),
- rtp_receiver_(NULL),
module_(NULL) {}
- void SetSendModule(RtpRtcp* rtpRtcpModule,
- RTPPayloadRegistry* rtp_payload_registry,
- RtpReceiver* receiver) {
+ void SetSendModule(RtpRtcp* rtpRtcpModule) {
module_ = rtpRtcpModule;
- rtp_payload_registry_ = rtp_payload_registry;
- rtp_receiver_ = receiver;
}
void DropEveryNthPacket(int n) { packet_loss_ = n; }
@@ -99,24 +78,15 @@
size_t len,
const PacketOptions& options) override {
count_++;
- const unsigned char* ptr = static_cast<const unsigned char*>(data);
- uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
- if (ssrc == rtx_ssrc_)
- count_rtx_ssrc_++;
- uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
- size_t packet_length = len;
- uint8_t restored_packet[1500];
- RTPHeader header;
- std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
- if (!parser->Parse(ptr, len, &header)) {
+ RtpPacketReceived packet;
+ if (!packet.Parse(data, len))
return false;
- }
-
- if (!rtp_payload_registry_->IsRtx(header)) {
- // Don't store retransmitted packets since we compare it to the list
- // created by the receiver.
+ if (packet.Ssrc() == rtx_ssrc_) {
+ count_rtx_ssrc_++;
+ } else {
+ // For non-RTX packets only.
expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
- sequence_number);
+ packet.SequenceNumber());
}
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
@@ -126,28 +96,7 @@
count_ < consecutive_drop_end_) {
return true;
}
- if (rtp_payload_registry_->IsRtx(header)) {
- // Remove the RTX header and parse the original RTP header.
- EXPECT_TRUE(rtp_payload_registry_->RestoreOriginalPacket(
- restored_packet, ptr, &packet_length, rtp_receiver_->SSRC(), header));
- if (!parser->Parse(restored_packet, packet_length, &header)) {
- return false;
- }
- ptr = restored_packet;
- } else {
- rtp_payload_registry_->SetIncomingPayloadType(header);
- }
-
- PayloadUnion payload_specific;
- if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
- &payload_specific)) {
- return false;
- }
- if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength,
- packet_length - header.headerLength,
- payload_specific, true)) {
- return false;
- }
+ EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
return true;
}
@@ -160,9 +109,8 @@
int consecutive_drop_end_;
uint32_t rtx_ssrc_;
int count_rtx_ssrc_;
- RTPPayloadRegistry* rtp_payload_registry_;
- RtpReceiver* rtp_receiver_;
RtpRtcp* module_;
+ RtpStreamReceiverController stream_receiver_controller_;
std::set<uint16_t> expected_sequence_numbers_;
};
@@ -170,8 +118,10 @@
protected:
RtpRtcpRtxNackTest()
: rtp_rtcp_module_(nullptr),
- transport_(kTestSsrc + 1),
- receiver_(),
+ transport_(kTestRtxSsrc),
+ rtx_stream_(&media_stream_,
+ rtx_associated_payload_types_,
+ kTestSsrc),
payload_data_length(sizeof(payload_data)),
fake_clock(123456),
retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
@@ -187,11 +137,6 @@
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
- rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
-
- rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
- &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
-
rtp_rtcp_module_->SetSSRC(kTestSsrc);
rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
@@ -199,18 +144,16 @@
rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
rtp_rtcp_module_->SetStartTimestamp(111111);
- transport_.SetSendModule(rtp_rtcp_module_, &rtp_payload_registry_,
- rtp_receiver_.get());
+ // Used for NACK processing.
+ // TODO(nisse): Unclear on which side? It's confusing to use a
+ // single rtp_rtcp module for both send and receive side.
+ rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);
- VideoCodec video_codec;
- memset(&video_codec, 0, sizeof(video_codec));
- video_codec.plType = kPayloadType;
- memcpy(video_codec.plName, "I420", 5);
-
- EXPECT_EQ(0, rtp_rtcp_module_->RegisterSendPayload(video_codec));
+ rtp_rtcp_module_->RegisterVideoSendPayload(kPayloadType, "video");
rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
- EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(video_codec));
- rtp_payload_registry_.SetRtxPayloadType(kRtxPayloadType, kPayloadType);
+ transport_.SetSendModule(rtp_rtcp_module_);
+ media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+ kTestSsrc, &media_stream_);
for (size_t n = 0; n < payload_data_length; n++) {
payload_data[n] = n % 10;
@@ -218,14 +161,14 @@
}
int BuildNackList(uint16_t* nack_list) {
- receiver_.sequence_numbers_.sort();
+ media_stream_.sequence_numbers_.sort();
std::list<uint16_t> missing_sequence_numbers;
- std::list<uint16_t>::iterator it = receiver_.sequence_numbers_.begin();
+ std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();
- while (it != receiver_.sequence_numbers_.end()) {
+ while (it != media_stream_.sequence_numbers_.end()) {
uint16_t sequence_number_1 = *it;
++it;
- if (it != receiver_.sequence_numbers_.end()) {
+ if (it != media_stream_.sequence_numbers_.end()) {
uint16_t sequence_number_2 = *it;
// Add all missing sequence numbers to list
for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
@@ -243,8 +186,8 @@
bool ExpectedPacketsReceived() {
std::list<uint16_t> received_sorted;
- std::copy(receiver_.sequence_numbers_.begin(),
- receiver_.sequence_numbers_.end(),
+ std::copy(media_stream_.sequence_numbers_.begin(),
+ media_stream_.sequence_numbers_.end(),
std::back_inserter(received_sorted));
received_sorted.sort();
return received_sorted.size() ==
@@ -254,9 +197,10 @@
}
void RunRtxTest(RtxMode rtx_method, int loss) {
- rtp_payload_registry_.SetRtxSsrc(kTestSsrc + 1);
+ rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+ kTestRtxSsrc, &rtx_stream_);
rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
- rtp_rtcp_module_->SetRtxSsrc(kTestSsrc + 1);
+ rtp_rtcp_module_->SetRtxSsrc(kTestRtxSsrc);
transport_.DropEveryNthPacket(loss);
uint32_t timestamp = 3000;
uint16_t nack_list[kVideoNackListSize];
@@ -274,22 +218,24 @@
// Prepare next frame.
timestamp += 3000;
}
- receiver_.sequence_numbers_.sort();
+ media_stream_.sequence_numbers_.sort();
}
void TearDown() override { delete rtp_rtcp_module_; }
std::unique_ptr<ReceiveStatistics> receive_statistics_;
- RTPPayloadRegistry rtp_payload_registry_;
- std::unique_ptr<RtpReceiver> rtp_receiver_;
RtpRtcp* rtp_rtcp_module_;
- std::unique_ptr<TestRtpFeedback> rtp_feedback_;
RtxLoopBackTransport transport_;
- VerifyingRtxReceiver receiver_;
+ const std::map<int, int> rtx_associated_payload_types_ =
+ {{kRtxPayloadType, kPayloadType}};
+ VerifyingMediaStream media_stream_;
+ RtxReceiveStream rtx_stream_;
uint8_t payload_data[65000];
size_t payload_data_length;
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
+ std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+ std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
};
TEST_F(RtpRtcpRtxNackTest, LongNackList) {
@@ -316,26 +262,26 @@
rtp_rtcp_module_->Process();
}
EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
- EXPECT_FALSE(receiver_.sequence_numbers_.empty());
- size_t last_receive_count = receiver_.sequence_numbers_.size();
+ EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
+ size_t last_receive_count = media_stream_.sequence_numbers_.size();
int length = BuildNackList(nack_list);
for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
rtp_rtcp_module_->SendNACK(nack_list, length);
- EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
- last_receive_count = receiver_.sequence_numbers_.size();
+ EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
+ last_receive_count = media_stream_.sequence_numbers_.size();
EXPECT_FALSE(ExpectedPacketsReceived());
}
rtp_rtcp_module_->SendNACK(nack_list, length);
- EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
+ EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
EXPECT_TRUE(ExpectedPacketsReceived());
}
TEST_F(RtpRtcpRtxNackTest, RtxNack) {
RunRtxTest(kRtxRetransmitted, 10);
- EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
+ EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(receiver_.sequence_numbers_.rbegin()));
- EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
+ *(media_stream_.sequence_numbers_.rbegin()));
+ EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
EXPECT_TRUE(ExpectedPacketsReceived());
}
diff --git a/modules/rtp_rtcp/source/rtp_payload_registry.cc b/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 35616b6..fe2bc80 100644
--- a/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -14,7 +14,6 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/stringutils.h"
@@ -270,65 +269,10 @@
return rtx_;
}
-bool RTPPayloadRegistry::IsRtx(const RTPHeader& header) const {
- rtc::CritScope cs(&crit_sect_);
- return IsRtxInternal(header);
-}
-
bool RTPPayloadRegistry::IsRtxInternal(const RTPHeader& header) const {
return rtx_ && ssrc_rtx_ == header.ssrc;
}
-bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
- const uint8_t* packet,
- size_t* packet_length,
- uint32_t original_ssrc,
- const RTPHeader& header) {
- if (kRtxHeaderSize + header.headerLength + header.paddingLength >
- *packet_length) {
- return false;
- }
- const uint8_t* rtx_header = packet + header.headerLength;
- uint16_t original_sequence_number = (rtx_header[0] << 8) + rtx_header[1];
-
- // Copy the packet into the restored packet, except for the RTX header.
- memcpy(restored_packet, packet, header.headerLength);
- memcpy(restored_packet + header.headerLength,
- packet + header.headerLength + kRtxHeaderSize,
- *packet_length - header.headerLength - kRtxHeaderSize);
- *packet_length -= kRtxHeaderSize;
-
- // Replace the SSRC and the sequence number with the originals.
- ByteWriter<uint16_t>::WriteBigEndian(restored_packet + 2,
- original_sequence_number);
- ByteWriter<uint32_t>::WriteBigEndian(restored_packet + 8, original_ssrc);
-
- rtc::CritScope cs(&crit_sect_);
- if (!rtx_)
- return true;
-
- auto apt_mapping = rtx_payload_type_map_.find(header.payloadType);
- if (apt_mapping == rtx_payload_type_map_.end()) {
- // No associated payload type found. Warn, unless we have already done so.
- if (payload_types_with_suppressed_warnings_.find(header.payloadType) ==
- payload_types_with_suppressed_warnings_.end()) {
- LOG(LS_WARNING)
- << "No RTX associated payload type mapping was available; "
- "not able to restore original packet from RTX packet "
- "with payload type: "
- << static_cast<int>(header.payloadType) << ". "
- << "Suppressing further warnings for this payload type.";
- payload_types_with_suppressed_warnings_.insert(header.payloadType);
- }
- return false;
- }
- restored_packet[1] = static_cast<uint8_t>(apt_mapping->second);
- if (header.markerBit) {
- restored_packet[1] |= kRtpMarkerBitMask; // Marker bit is set.
- }
- return true;
-}
-
void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope cs(&crit_sect_);
ssrc_rtx_ = ssrc;
@@ -359,10 +303,6 @@
return it != payload_type_map_.end() && _stricmp(it->second.name, "red") == 0;
}
-bool RTPPayloadRegistry::IsEncapsulated(const RTPHeader& header) const {
- return IsRed(header) || IsRtx(header);
-}
-
bool RTPPayloadRegistry::GetPayloadSpecifics(uint8_t payload_type,
PayloadUnion* payload) const {
rtc::CritScope cs(&crit_sect_);
diff --git a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index f5707d2..6e17eb9 100644
--- a/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -13,7 +13,6 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
@@ -259,105 +258,6 @@
rtp_payload_registry.RegisterReceivePayload(audio_codec, &ignored));
}
-// Generates an RTX packet for the given length and original sequence number.
-// The RTX sequence number and ssrc will use the default value of 9999. The
-// caller takes ownership of the returned buffer.
-const uint8_t* GenerateRtxPacket(size_t header_length,
- size_t payload_length,
- uint16_t original_sequence_number) {
- uint8_t* packet =
- new uint8_t[kRtxHeaderSize + header_length + payload_length]();
- // Write the RTP version to the first byte, so the resulting header can be
- // parsed.
- static const int kRtpExpectedVersion = 2;
- packet[0] = static_cast<uint8_t>(kRtpExpectedVersion << 6);
- // Write a junk sequence number. It should be thrown away when the packet is
- // restored.
- ByteWriter<uint16_t>::WriteBigEndian(packet + 2, 9999);
- // Write a junk ssrc. It should also be thrown away when the packet is
- // restored.
- ByteWriter<uint32_t>::WriteBigEndian(packet + 8, 9999);
-
- // Now write the RTX header. It occurs at the start of the payload block, and
- // contains just the sequence number.
- ByteWriter<uint16_t>::WriteBigEndian(packet + header_length,
- original_sequence_number);
- return packet;
-}
-
-void TestRtxPacket(RTPPayloadRegistry* rtp_payload_registry,
- int rtx_payload_type,
- int expected_payload_type,
- bool should_succeed) {
- size_t header_length = 100;
- size_t payload_length = 200;
- size_t original_length = header_length + payload_length + kRtxHeaderSize;
-
- RTPHeader header;
- header.ssrc = 1000;
- header.sequenceNumber = 100;
- header.payloadType = rtx_payload_type;
- header.headerLength = header_length;
-
- uint16_t original_sequence_number = 1234;
- uint32_t original_ssrc = 500;
-
- std::unique_ptr<const uint8_t[]> packet(GenerateRtxPacket(
- header_length, payload_length, original_sequence_number));
- std::unique_ptr<uint8_t[]> restored_packet(
- new uint8_t[header_length + payload_length]);
- size_t length = original_length;
- bool success = rtp_payload_registry->RestoreOriginalPacket(
- restored_packet.get(), packet.get(), &length, original_ssrc, header);
- EXPECT_EQ(should_succeed, success)
- << "Test success should match should_succeed.";
- if (!success) {
- return;
- }
-
- EXPECT_EQ(original_length - kRtxHeaderSize, length)
- << "The restored packet should be exactly kRtxHeaderSize smaller.";
-
- std::unique_ptr<RtpHeaderParser> header_parser(RtpHeaderParser::Create());
- RTPHeader restored_header;
- ASSERT_TRUE(
- header_parser->Parse(restored_packet.get(), length, &restored_header));
- EXPECT_EQ(original_sequence_number, restored_header.sequenceNumber)
- << "The restored packet should have the original sequence number "
- << "in the correct location in the RTP header.";
- EXPECT_EQ(expected_payload_type, restored_header.payloadType)
- << "The restored packet should have the correct payload type.";
- EXPECT_EQ(original_ssrc, restored_header.ssrc)
- << "The restored packet should have the correct ssrc.";
-}
-
-TEST(RtpPayloadRegistryTest, MultipleRtxPayloadTypes) {
- RTPPayloadRegistry rtp_payload_registry;
- // Set the incoming payload type to 90.
- RTPHeader header;
- header.payloadType = 90;
- header.ssrc = 1;
- rtp_payload_registry.SetIncomingPayloadType(header);
- rtp_payload_registry.SetRtxSsrc(100);
- // Map two RTX payload types.
- rtp_payload_registry.SetRtxPayloadType(105, 95);
- rtp_payload_registry.SetRtxPayloadType(106, 96);
-
- TestRtxPacket(&rtp_payload_registry, 105, 95, true);
- TestRtxPacket(&rtp_payload_registry, 106, 96, true);
-}
-
-TEST(RtpPayloadRegistryTest, InvalidRtxConfiguration) {
- RTPPayloadRegistry rtp_payload_registry;
- rtp_payload_registry.SetRtxSsrc(100);
- // Fails because no mappings exist and the incoming payload type isn't known.
- TestRtxPacket(&rtp_payload_registry, 105, 0, false);
- // Succeeds when the mapping is used, but fails for the implicit fallback.
- rtp_payload_registry.SetRtxPayloadType(105, 95);
- TestRtxPacket(&rtp_payload_registry, 105, 95, true);
- TestRtxPacket(&rtp_payload_registry, 106, 0, false);
-}
-
INSTANTIATE_TEST_CASE_P(TestDynamicRange,
RtpPayloadRegistryGenericTest,
testing::Range(96, 127 + 1));
diff --git a/modules/rtp_rtcp/test/testAPI/test_api.cc b/modules/rtp_rtcp/test/testAPI/test_api.cc
index b39c8d7..28be222 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -162,23 +162,4 @@
EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
}
-TEST_F(RtpRtcpAPITest, RtxReceiver) {
- const uint32_t kRtxSsrc = 1;
- const int kRtxPayloadType = 119;
- const int kPayloadType = 100;
- EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
- rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
- rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType, kPayloadType);
- EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
- RTPHeader rtx_header;
- rtx_header.ssrc = kRtxSsrc;
- rtx_header.payloadType = kRtxPayloadType;
- EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
- rtx_header.ssrc = 0;
- EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
- rtx_header.ssrc = kRtxSsrc;
- rtx_header.payloadType = 0;
- EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
-}
-
} // namespace webrtc
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 1dbd869..5aff226 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -453,7 +453,7 @@
size_t packet_length,
const RTPHeader& header,
bool in_order) {
- if (rtp_payload_registry_.IsEncapsulated(header)) {
+ if (rtp_payload_registry_.IsRed(header)) {
ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
return;
}
@@ -485,8 +485,6 @@
return;
}
ulpfec_receiver_->ProcessReceivedFec();
- } else if (rtp_payload_registry_.IsRtx(header)) {
- LOG(LS_WARNING) << "Unexpected RTX packet on media ssrc";
}
}
