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}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/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 e46440b..f432f0a 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(
