Reland "Delete test/constants.h"
This reverts commit 4f36b7a478c2763463c7a9ea970548ec68bc3ea6.
Reason for revert: Failing tests fixed.
Original change's description:
> Revert "Delete test/constants.h"
>
> This reverts commit 389b1672a32f2dd49af6c6ed40e8ddf394b986de.
>
> Reason for revert: Causes failure (and empty result list) in CallPerfTest.PadsToMinTransmitBitrate
>
> Original change's description:
> > Delete test/constants.h
> >
> > It's not possible to use constants.h for all RTP extensions
> > after the number of extensions exceeds 14, which is the maximum
> > number of one-byte RTP extensions. This is because some extensions
> > would have to be assigned a number greater than 14, even if the
> > test only involves 14 extensions or less.
> >
> > For uniformity's sake, this CL also edits some files to use an
> > enum as the files involved in this CL, rather than free-floating
> > const-ints.
> >
> > Bug: webrtc:10288
> > Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123048
> > Commit-Queue: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26728}
>
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
>
> Bug: webrtc:10288, chromium:933127
> Change-Id: If1de0bd8992137c52bf0b877b3cb0a2bafc809d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/123381
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26744}
TBR=danilchap@webrtc.org,oprypin@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org
Change-Id: I65e391325d3a6df6db3c0739185e2002e70fb954
Bug: webrtc:10288, chromium:933127
Reviewed-on: https://webrtc-review.googlesource.com/c/123384
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26750}diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
index 55de03d..8a83554 100644
--- a/audio/audio_send_stream_tests.cc
+++ b/audio/audio_send_stream_tests.cc
@@ -8,8 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
+#include <utility>
+#include <vector>
+
#include "test/call_test.h"
-#include "test/constants.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
@@ -18,6 +21,11 @@
namespace test {
namespace {
+enum : int { // The first valid value is 1.
+ kAudioLevelExtensionId = 1,
+ kTransportSequenceNumberExtensionId,
+};
+
class AudioSendTest : public SendTest {
public:
AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
@@ -103,8 +111,8 @@
class AudioLevelObserver : public AudioSendTest {
public:
AudioLevelObserver() : AudioSendTest() {
- EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, test::kAudioLevelExtensionId));
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
+ kAudioLevelExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -127,8 +135,8 @@
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAudioLevelUri, test::kAudioLevelExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelExtensionId));
}
void PerformTest() override {
diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc
index 74eaef0..2e3c158 100644
--- a/audio/test/audio_bwe_integration_test.cc
+++ b/audio/test/audio_bwe_integration_test.cc
@@ -23,6 +23,10 @@
namespace test {
namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+};
+
// Wait a second between stopping sending and stopping receiving audio.
constexpr int kExtraProcessTimeMs = 1000;
} // namespace
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 3ffcb6b..7325c15 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -51,8 +51,19 @@
using webrtc::test::DriftingClock;
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+};
+} // namespace
class CallPerfTest : public test::CallTest {
+ public:
+ CallPerfTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ }
+
protected:
enum class FecMode { kOn, kOff };
enum class CreateOrder { kAudioFirst, kVideoFirst };
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 7f9d72f..c5ca7f0 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -22,7 +22,10 @@
namespace webrtc {
namespace {
-const int kAudioLevelExtensionId = 9;
+enum : int { // The first valid value is 1.
+ kAudioLevelExtensionId = 1,
+};
+
const uint16_t kSeqNum = 33;
const uint32_t kSsrc = 725242;
const uint8_t kAudioLevel = 0x5a;
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 714fac7..0b881f8 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -40,16 +40,19 @@
namespace webrtc {
namespace {
-const int kTransmissionTimeOffsetExtensionId = 1;
-const int kAbsoluteSendTimeExtensionId = 14;
-const int kTransportSequenceNumberExtensionId = 13;
-const int kVideoTimingExtensionId = 12;
-const int kMidExtensionId = 11;
-const int kGenericDescriptorId = 10;
-const int kAudioLevelExtensionId = 9;
-const int kRidExtensionId = 8;
-const int kRepairedRidExtensionId = 7;
-const int kVideoRotationExtensionId = 5;
+enum : int { // The first valid value is 1.
+ kAbsoluteSendTimeExtensionId = 1,
+ kAudioLevelExtensionId,
+ kGenericDescriptorId,
+ kMidExtensionId,
+ kRepairedRidExtensionId,
+ kRidExtensionId,
+ kTransmissionTimeOffsetExtensionId,
+ kTransportSequenceNumberExtensionId,
+ kVideoRotationExtensionId,
+ kVideoTimingExtensionId,
+};
+
const int kPayload = 100;
const int kRtxPayload = 98;
const uint32_t kTimestamp = 10;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 43eb4ed..0cd0572 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -34,13 +34,16 @@
using ::testing::ElementsAre;
-constexpr int kTransmissionTimeOffsetExtensionId = 1;
-constexpr int kAbsoluteSendTimeExtensionId = 14;
-constexpr int kTransportSequenceNumberExtensionId = 13;
-constexpr int kVideoTimingExtensionId = 12;
-constexpr int kGenericDescriptorId = 10;
-constexpr int kFrameMarkingExtensionId = 6;
-constexpr int kVideoRotationExtensionId = 5;
+enum : int { // The first valid value is 1.
+ kAbsoluteSendTimeExtensionId = 1,
+ kFrameMarkingExtensionId,
+ kGenericDescriptorId,
+ kTransmissionTimeOffsetExtensionId,
+ kTransportSequenceNumberExtensionId,
+ kVideoRotationExtensionId,
+ kVideoTimingExtensionId,
+};
+
constexpr int kPayload = 100;
constexpr uint32_t kTimestamp = 10;
constexpr uint16_t kSeqNum = 33;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 99c5467..8acc26d 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -656,8 +656,6 @@
sources = [
"call_test.cc",
"call_test.h",
- "constants.cc",
- "constants.h",
"drifting_clock.cc",
"drifting_clock.h",
"encoder_settings.cc",
@@ -748,6 +746,7 @@
"../video",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
+ "//third_party/abseil-cpp/absl/types:optional",
]
if (!is_android && !build_with_chromium) {
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
diff --git a/test/call_test.cc b/test/call_test.cc
index 57a5d9f..ca0d7bb 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -65,6 +65,29 @@
});
}
+void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
+ for (const RtpExtension& registered_extension : rtp_extensions_) {
+ if (registered_extension.id == extension.id) {
+ ASSERT_EQ(registered_extension.uri, extension.uri)
+ << "Different URIs associated with ID " << extension.id << ".";
+ ASSERT_EQ(registered_extension.encrypt, extension.encrypt)
+ << "Encryption mismatch associated with ID " << extension.id << ".";
+ return;
+ } else { // Different IDs.
+ // Different IDs referring to the same extension probably indicate
+ // a mistake in the test.
+ ASSERT_FALSE(registered_extension.uri == extension.uri &&
+ registered_extension.encrypt == extension.encrypt)
+ << "URI " << extension.uri
+ << (extension.encrypt ? " with " : " without ")
+ << "encryption already registered with a different "
+ << "ID (" << extension.id << " vs. " << registered_extension.id
+ << ").";
+ }
+ }
+ rtp_extensions_.push_back(extension);
+}
+
void CallTest::RunBaseTest(BaseTest* test) {
task_queue_.SendTask([this, test]() {
num_video_streams_ = test->GetNumVideoStreams();
@@ -235,25 +258,23 @@
video_config->rtp.payload_name = "FAKE";
video_config->rtp.payload_type = kFakeVideoSendPayloadType;
video_config->rtp.extmap_allow_mixed = true;
- video_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
- video_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
- video_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kGenericFrameDescriptorUri, kGenericDescriptorExtensionId));
+ AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri,
+ &video_config->rtp.extensions);
if (video_encoder_configs_.empty()) {
video_encoder_configs_.emplace_back();
FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
&video_encoder_configs_.back());
}
-
for (size_t i = 0; i < num_video_streams; ++i)
video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
- video_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
- video_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
+ AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
+ &video_config->rtp.extensions);
}
void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
@@ -666,6 +687,25 @@
return &flexfec_receive_configs_[0];
}
+absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
+ const std::string& uri) const {
+ for (const auto& extension : rtp_extensions_) {
+ if (extension.uri == uri) {
+ return extension;
+ }
+ }
+ return absl::nullopt;
+}
+
+void CallTest::AddRtpExtensionByUri(
+ const std::string& uri,
+ std::vector<RtpExtension>* extensions) const {
+ const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
+ if (extension) {
+ extensions->push_back(*extension);
+ }
+}
+
constexpr size_t CallTest::kNumSsrcs;
const int CallTest::kDefaultWidth;
const int CallTest::kDefaultHeight;
diff --git a/test/call_test.h b/test/call_test.h
index fb26051..dbe8e07 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -15,6 +15,7 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/video_bitrate_allocator_factory.h"
@@ -71,6 +72,8 @@
static const std::map<uint8_t, MediaType> payload_type_map_;
protected:
+ void RegisterRtpExtension(const RtpExtension& extension);
+
// RunBaseTest overwrites the audio_state of the send and receive Call configs
// to simplify test code.
void RunBaseTest(BaseTest* test);
@@ -216,6 +219,13 @@
SingleThreadedTaskQueueForTesting task_queue_;
private:
+ absl::optional<RtpExtension> GetRtpExtensionByUri(
+ const std::string& uri) const;
+
+ void AddRtpExtensionByUri(const std::string& uri,
+ std::vector<RtpExtension>* extensions) const;
+
+ std::vector<RtpExtension> rtp_extensions_;
rtc::scoped_refptr<AudioProcessing> apm_send_;
rtc::scoped_refptr<AudioProcessing> apm_recv_;
rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
diff --git a/test/constants.cc b/test/constants.cc
deleted file mode 100644
index 4f33d25..0000000
--- a/test/constants.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (c) 2015 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 "test/constants.h"
-
-namespace webrtc {
-namespace test {
-
-const int kAudioLevelExtensionId = 5;
-const int kTOffsetExtensionId = 6;
-const int kAbsSendTimeExtensionId = 7;
-const int kTransportSequenceNumberExtensionId = 8;
-const int kVideoRotationExtensionId = 9;
-const int kVideoContentTypeExtensionId = 10;
-const int kVideoTimingExtensionId = 11;
-const int kGenericDescriptorExtensionId = 12;
-const int kColorSpaceExtensionId = 13;
-
-} // namespace test
-} // namespace webrtc
diff --git a/test/constants.h b/test/constants.h
deleted file mode 100644
index b1b87d6..0000000
--- a/test/constants.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- * Copyright (c) 2015 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.
- */
-
-namespace webrtc {
-namespace test {
-
-extern const int kAudioLevelExtensionId;
-extern const int kTOffsetExtensionId;
-extern const int kAbsSendTimeExtensionId;
-extern const int kTransportSequenceNumberExtensionId;
-extern const int kVideoRotationExtensionId;
-extern const int kVideoContentTypeExtensionId;
-extern const int kVideoTimingExtensionId;
-extern const int kGenericDescriptorExtensionId;
-extern const int kColorSpaceExtensionId;
-} // namespace test
-} // namespace webrtc
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index 7cff645..830c2f1 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -12,6 +12,7 @@
#include <map>
#include <memory>
+#include <utility>
#include <vector>
#include "api/test/simulated_network.h"
@@ -21,7 +22,6 @@
#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
#include "system_wrappers/include/field_trial.h"
-#include "test/constants.h"
#include "test/direct_transport.h"
#include "test/gtest.h"
@@ -71,14 +71,7 @@
protected:
RtpRtcpObserver() : RtpRtcpObserver(0) {}
explicit RtpRtcpObserver(int event_timeout_ms)
- : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {
- parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
- kTOffsetExtensionId);
- parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
- kAbsSendTimeExtensionId);
- parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
- }
+ : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {}
rtc::Event observation_complete_;
const std::unique_ptr<RtpHeaderParser> parser_;
diff --git a/test/scenario/video_stream.cc b/test/scenario/video_stream.cc
index 2f892f0..859626e 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -27,8 +27,13 @@
namespace webrtc {
namespace test {
namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+ kVideoContentTypeExtensionId,
+ kVideoRotationRtpExtensionId,
+};
+
constexpr int kDefaultMaxQp = cricket::WebRtcVideoChannel::kDefaultQpMax;
-const int kVideoRotationRtpExtensionId = 4;
uint8_t CodecTypeToPayloadType(VideoCodecType codec_type) {
switch (codec_type) {
case VideoCodecType::kVideoCodecGeneric:
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 17bb985..e7550a4 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -26,6 +26,11 @@
#include "test/video_encoder_proxy_factory.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kAbsSendTimeExtensionId = 1,
+};
+} // namespace
class BandwidthEndToEndTest : public test::CallTest {
public:
@@ -42,8 +47,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
@@ -87,8 +92,8 @@
VideoEncoderConfig* encoder_config) override {
if (!send_side_bwe_) {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
diff --git a/video/end_to_end_tests/codec_tests.cc b/video/end_to_end_tests/codec_tests.cc
index ed303ec..eab9ef7 100644
--- a/video/end_to_end_tests/codec_tests.cc
+++ b/video/end_to_end_tests/codec_tests.cc
@@ -27,11 +27,22 @@
#include "test/gtest.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kColorSpaceExtensionId = 1,
+ kVideoRotationExtensionId,
+};
+} // namespace
class CodecEndToEndTest : public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
- CodecEndToEndTest() : field_trial_(GetParam()) {}
+ CodecEndToEndTest() : field_trial_(GetParam()) {
+ RegisterRtpExtension(
+ RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+ kVideoRotationExtensionId));
+ }
private:
test::ScopedFieldTrials field_trial_;
@@ -224,7 +235,10 @@
class EndToEndTestH264 : public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
- EndToEndTestH264() : field_trial_(GetParam()) {}
+ EndToEndTestH264() : field_trial_(GetParam()) {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+ kVideoRotationExtensionId));
+ }
private:
test::ScopedFieldTrials field_trial_;
diff --git a/video/end_to_end_tests/extended_reports_tests.cc b/video/end_to_end_tests/extended_reports_tests.cc
index 5b649d5..955f9cd 100644
--- a/video/end_to_end_tests/extended_reports_tests.cc
+++ b/video/end_to_end_tests/extended_reports_tests.cc
@@ -43,8 +43,20 @@
#include "test/single_threaded_task_queue.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kColorSpaceExtensionId = 1,
+ kTransportSequenceNumberExtensionId,
+};
+} // namespace
-class ExtendedReportsEndToEndTest : public test::CallTest {};
+class ExtendedReportsEndToEndTest : public test::CallTest {
+ public:
+ ExtendedReportsEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ }
+};
class RtcpXrObserver : public test::EndToEndTest {
public:
diff --git a/video/end_to_end_tests/fec_tests.cc b/video/end_to_end_tests/fec_tests.cc
index 42215d3..dbf0b2c 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -22,10 +22,21 @@
#include "test/rtcp_packet_parser.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+ kVideoRotationExtensionId,
+};
+} // namespace
class FecEndToEndTest : public test::CallTest {
public:
- FecEndToEndTest() = default;
+ FecEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+ kVideoRotationExtensionId));
+ }
};
TEST_F(FecEndToEndTest, ReceivesUlpfec) {
diff --git a/video/end_to_end_tests/frame_encryption_tests.cc b/video/end_to_end_tests/frame_encryption_tests.cc
index 98b3a4b..85ad7dd 100644
--- a/video/end_to_end_tests/frame_encryption_tests.cc
+++ b/video/end_to_end_tests/frame_encryption_tests.cc
@@ -17,10 +17,18 @@
#include "test/gtest.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kGenericDescriptorExtensionId = 1,
+};
+} // namespace
class FrameEncryptionEndToEndTest : public test::CallTest {
public:
- FrameEncryptionEndToEndTest() = default;
+ FrameEncryptionEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kGenericFrameDescriptorUri,
+ kGenericDescriptorExtensionId));
+ }
private:
// GenericDescriptor is required for FrameEncryption to work.
diff --git a/video/end_to_end_tests/histogram_tests.cc b/video/end_to_end_tests/histogram_tests.cc
index 312924c..ef435b4 100644
--- a/video/end_to_end_tests/histogram_tests.cc
+++ b/video/end_to_end_tests/histogram_tests.cc
@@ -16,8 +16,22 @@
#include "test/gtest.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+ kVideoContentTypeExtensionId,
+};
+} // namespace
class HistogramTest : public test::CallTest {
+ public:
+ HistogramTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
+ kVideoContentTypeExtensionId));
+ }
+
protected:
void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
};
diff --git a/video/end_to_end_tests/probing_tests.cc b/video/end_to_end_tests/probing_tests.cc
index 80c9ebd..dba3c3e 100644
--- a/video/end_to_end_tests/probing_tests.cc
+++ b/video/end_to_end_tests/probing_tests.cc
@@ -15,9 +15,21 @@
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/gtest.h"
-namespace webrtc {
-class ProbingEndToEndTest : public test::CallTest {};
+namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+};
+} // namespace
+
+class ProbingEndToEndTest : public test::CallTest {
+ public:
+ ProbingEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ }
+};
class ProbingTest : public test::EndToEndTest {
public:
diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc
index 7d31aed..49c3fb4 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -21,9 +21,18 @@
#include "test/rtcp_packet_parser.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kVideoRotationExtensionId = 1,
+};
+} // namespace
+
class RetransmissionEndToEndTest : public test::CallTest {
public:
- RetransmissionEndToEndTest() = default;
+ RetransmissionEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
+ kVideoRotationExtensionId));
+ }
protected:
void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
diff --git a/video/end_to_end_tests/rtp_rtcp_tests.cc b/video/end_to_end_tests/rtp_rtcp_tests.cc
index bef15d2..d89a024 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -18,6 +18,11 @@
#include "test/rtcp_packet_parser.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+};
+} // namespace
class RtpRtcpEndToEndTest : public test::CallTest {
protected:
@@ -530,7 +535,7 @@
flexfec_receive_config.transport_cc = true;
flexfec_receive_config.rtp_header_extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
- test::kTransportSequenceNumberExtensionId);
+ kTransportSequenceNumberExtensionId);
flexfec_receive_configs_.push_back(flexfec_receive_config);
CreateFlexfecStreams();
diff --git a/video/end_to_end_tests/stats_tests.cc b/video/end_to_end_tests/stats_tests.cc
index 53b181a..bad04b4 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -24,7 +24,19 @@
#include "test/rtcp_packet_parser.h"
namespace webrtc {
-class StatsEndToEndTest : public test::CallTest {};
+namespace {
+enum : int { // The first valid value is 1.
+ kVideoContentTypeExtensionId = 1,
+};
+} // namespace
+
+class StatsEndToEndTest : public test::CallTest {
+ public:
+ StatsEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
+ kVideoContentTypeExtensionId));
+ }
+};
TEST_F(StatsEndToEndTest, GetStats) {
static const int kStartBitrateBps = 3000000;
diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc
index 9aacd9a..b607bf6 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -21,12 +21,21 @@
#include "video/end_to_end_tests/multi_stream_tester.h"
namespace webrtc {
+namespace {
+enum : int { // The first valid value is 1.
+ kTransportSequenceNumberExtensionId = 1,
+};
+} // namespace
-class TransportFeedbackEndToEndTest : public test::CallTest {};
+class TransportFeedbackEndToEndTest : public test::CallTest {
+ public:
+ TransportFeedbackEndToEndTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ }
+};
TEST_F(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
- static const int kExtensionId = 5;
-
class RtpExtensionHeaderObserver : public test::DirectTransport {
public:
RtpExtensionHeaderObserver(
@@ -50,7 +59,7 @@
retransmit_observed_(false),
started_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kExtensionId);
+ kTransportSequenceNumberExtensionId);
}
virtual ~RtpExtensionHeaderObserver() {}
@@ -174,8 +183,9 @@
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
// Force some padding to be sent. Note that since we do send media
// packets we can not guarantee that a padding only packet is sent.
@@ -201,8 +211,9 @@
VideoReceiveStream::Config* receive_config) override {
receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
receive_config->rtp.extensions.clear();
- receive_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ receive_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
receive_config->renderer = &fake_renderer_;
}
@@ -291,14 +302,14 @@
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
(*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
}
private:
- static const int kExtensionId = 5;
const bool feedback_enabled_;
const size_t num_video_streams_;
const size_t num_audio_streams_;
@@ -426,7 +437,6 @@
TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) {
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
- static constexpr int kExtensionId = 8;
static constexpr size_t kMinPacketsToWaitFor = 50;
class TransportSequenceNumberTest : public test::EndToEndTest {
public:
@@ -435,7 +445,7 @@
video_observed_(false),
audio_observed_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kExtensionId);
+ kTransportSequenceNumberExtensionId);
}
size_t GetNumVideoStreams() const override { return 1; }
@@ -445,8 +455,9 @@
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
}
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index ad3224d..4941827 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -47,6 +47,14 @@
namespace webrtc {
namespace {
+enum : int { // The first valid value is 1.
+ kAbsSendTimeExtensionId = 1,
+ kGenericFrameDescriptorExtensionId,
+ kTransportSequenceNumberExtensionId,
+ kVideoContentTypeExtensionId,
+ kVideoTimingExtensionId,
+};
+
constexpr char kSyncGroup[] = "av_sync";
constexpr int kOpusMinBitrateBps = 6000;
constexpr int kOpusBitrateFbBps = 32000;
@@ -710,10 +718,10 @@
if (params_.call.send_side_bwe) {
video_send_configs_[video_idx].rtp.extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
- test::kTransportSequenceNumberExtensionId);
+ kTransportSequenceNumberExtensionId);
} else {
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId);
+ RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId);
}
if (params_.call.generic_descriptor) {
@@ -724,13 +732,13 @@
video_send_configs_[video_idx].rtp.extensions.emplace_back(
RtpExtension::kGenericFrameDescriptorUri,
- test::kGenericDescriptorExtensionId);
+ kGenericFrameDescriptorExtensionId);
}
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId);
+ RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId);
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId);
+ RtpExtension::kVideoTimingUri, kVideoTimingExtensionId);
video_encoder_configs_[video_idx].video_format.name =
params_.video[video_idx].codec;
@@ -878,10 +886,10 @@
if (params_.call.send_side_bwe) {
GetFlexFecConfig()->rtp_header_extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- test::kTransportSequenceNumberExtensionId));
+ kTransportSequenceNumberExtensionId));
} else {
- GetFlexFecConfig()->rtp_header_extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ GetFlexFecConfig()->rtp_header_extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
}
@@ -919,10 +927,10 @@
if (params_.call.send_side_bwe) {
thumbnail_send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- test::kTransportSequenceNumberExtensionId));
+ kTransportSequenceNumberExtensionId));
} else {
- thumbnail_send_config.rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ thumbnail_send_config.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
VideoEncoderConfig thumbnail_encoder_config;
@@ -1354,7 +1362,7 @@
if (params_.call.send_side_bwe) {
audio_send_config.rtp.extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
- test::kTransportSequenceNumberExtensionId));
+ kTransportSequenceNumberExtensionId));
audio_send_config.min_bitrate_bps = kOpusMinBitrateBps;
audio_send_config.max_bitrate_bps = kOpusBitrateFbBps;
audio_send_config.send_codec_spec->transport_cc_enabled = true;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 057fd95..0beefec 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -73,6 +73,15 @@
} // namespace test
namespace {
+enum : int { // The first valid value is 1.
+ kAbsSendTimeExtensionId = 1,
+ kTimestampOffsetExtensionId,
+ kTransportSequenceNumberExtensionId,
+ kVideoContentTypeExtensionId,
+ kVideoRotationExtensionId,
+ kVideoTimingExtensionId,
+};
+
constexpr int64_t kRtcpIntervalMs = 1000;
enum VideoFormat {
@@ -84,6 +93,12 @@
VideoFrame CreateVideoFrame(int width, int height, uint8_t data);
class VideoSendStreamTest : public test::CallTest {
+ public:
+ VideoSendStreamTest() {
+ RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberExtensionId));
+ }
+
protected:
void TestNackRetransmission(uint32_t retransmit_ssrc,
uint8_t retransmit_payload_type);
@@ -163,7 +178,7 @@
public:
AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, test::kAbsSendTimeExtensionId));
+ kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -193,8 +208,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
void PerformTest() override {
@@ -215,7 +230,7 @@
Clock::GetRealTimeClock(), kEncodeDelayMs);
}) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset, kTimestampOffsetExtensionId));
}
private:
@@ -239,7 +254,7 @@
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
+ RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
}
void PerformTest() override {
@@ -253,7 +268,7 @@
}
TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
- static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+ static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
class TransportWideSequenceNumberObserver : public test::SendTest {
public:
TransportWideSequenceNumberObserver()
@@ -301,7 +316,7 @@
public:
VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoRotation, test::kVideoRotationExtensionId));
+ kRtpExtensionVideoRotation, kVideoRotationExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -322,7 +337,7 @@
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kVideoRotationUri, test::kVideoRotationExtensionId));
+ RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
}
void OnFrameGeneratorCapturerCreated(
@@ -344,7 +359,7 @@
VideoContentTypeObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoContentType, test::kVideoContentTypeExtensionId));
+ kRtpExtensionVideoContentType, kVideoContentTypeExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -367,9 +382,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kVideoContentTypeUri,
- test::kVideoContentTypeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
@@ -389,8 +403,8 @@
public:
VideoTimingObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
- EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoTiming, test::kVideoTimingExtensionId));
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionVideoTiming,
+ kVideoTimingExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -412,8 +426,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kVideoTimingUri, kVideoTimingExtensionId));
}
void PerformTest() override {
@@ -464,7 +478,12 @@
expect_ulpfec_(expect_ulpfec),
sent_media_(false),
sent_ulpfec_(false),
- header_extensions_enabled_(header_extensions_enabled) {}
+ header_extensions_enabled_(header_extensions_enabled) {
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsSendTimeExtensionId);
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+ kTransportSequenceNumberExtensionId);
+ }
// Some of the test cases are expected to time out and thus we are using
// a shorter timeout window than the default here.
@@ -564,12 +583,11 @@
VideoSendStreamTest::kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type =
VideoSendStreamTest::kUlpfecPayloadType;
- EXPECT_FALSE(send_config->rtp.extensions.empty());
if (!header_extensions_enabled_) {
send_config->rtp.extensions.clear();
} else {
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
}
encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
(*receive_configs)[0].rtp.red_payload_type =
@@ -687,7 +705,14 @@
sent_media_(false),
sent_flexfec_(false),
header_extensions_enabled_(header_extensions_enabled),
- num_video_streams_(num_video_streams) {}
+ num_video_streams_(num_video_streams) {
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsSendTimeExtensionId);
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
+ kTimestampOffsetExtensionId);
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+ kTransportSequenceNumberExtensionId);
+ }
size_t GetNumFlexfecStreams() const override { return 1; }
size_t GetNumVideoStreams() const override { return num_video_streams_; }
@@ -751,10 +776,10 @@
send_config->encoder_settings.encoder_factory = encoder_factory_;
send_config->rtp.payload_name = payload_name_;
if (header_extensions_enabled_) {
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
+ RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
} else {
send_config->rtp.extensions.clear();
}
@@ -1628,7 +1653,7 @@
TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
static const int kStartBitrateBps = 300000;
static const int kNewMaxBitrateBps = 1234567;
- static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
+ static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
class ChangingNetworkRouteTest : public test::EndToEndTest {
public:
explicit ChangingNetworkRouteTest(
