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
No-Presubmit: True
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}diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc
index 8a83554..55de03d 100644
--- a/audio/audio_send_stream_tests.cc
+++ b/audio/audio_send_stream_tests.cc
@@ -8,11 +8,8 @@
* 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"
@@ -21,11 +18,6 @@
namespace test {
namespace {
-enum : int { // The first valid value is 1.
- kAudioLevelExtensionId = 1,
- kTransportSequenceNumberExtensionId,
-};
-
class AudioSendTest : public SendTest {
public:
AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
@@ -111,8 +103,8 @@
class AudioLevelObserver : public AudioSendTest {
public:
AudioLevelObserver() : AudioSendTest() {
- EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
- kAudioLevelExtensionId));
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, test::kAudioLevelExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -135,8 +127,8 @@
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, kAudioLevelExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAudioLevelUri, test::kAudioLevelExtensionId));
}
void PerformTest() override {
diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc
index 2e3c158..74eaef0 100644
--- a/audio/test/audio_bwe_integration_test.cc
+++ b/audio/test/audio_bwe_integration_test.cc
@@ -23,10 +23,6 @@
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 c5ca7f0..7f9d72f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -22,10 +22,7 @@
namespace webrtc {
namespace {
-enum : int { // The first valid value is 1.
- kAudioLevelExtensionId = 1,
-};
-
+const int kAudioLevelExtensionId = 9;
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 0b881f8..714fac7 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -40,19 +40,16 @@
namespace webrtc {
namespace {
-enum : int { // The first valid value is 1.
- kAbsoluteSendTimeExtensionId = 1,
- kAudioLevelExtensionId,
- kGenericDescriptorId,
- kMidExtensionId,
- kRepairedRidExtensionId,
- kRidExtensionId,
- kTransmissionTimeOffsetExtensionId,
- kTransportSequenceNumberExtensionId,
- kVideoRotationExtensionId,
- kVideoTimingExtensionId,
-};
-
+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;
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 0cd0572..43eb4ed 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -34,16 +34,13 @@
using ::testing::ElementsAre;
-enum : int { // The first valid value is 1.
- kAbsoluteSendTimeExtensionId = 1,
- kFrameMarkingExtensionId,
- kGenericDescriptorId,
- kTransmissionTimeOffsetExtensionId,
- kTransportSequenceNumberExtensionId,
- kVideoRotationExtensionId,
- kVideoTimingExtensionId,
-};
-
+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;
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 8acc26d..99c5467 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -656,6 +656,8 @@
sources = [
"call_test.cc",
"call_test.h",
+ "constants.cc",
+ "constants.h",
"drifting_clock.cc",
"drifting_clock.h",
"encoder_settings.cc",
@@ -746,7 +748,6 @@
"../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 ca0d7bb..57a5d9f 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -65,29 +65,6 @@
});
}
-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();
@@ -258,23 +235,25 @@
video_config->rtp.payload_name = "FAKE";
video_config->rtp.payload_type = kFakeVideoSendPayloadType;
video_config->rtp.extmap_allow_mixed = true;
- AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
- &video_config->rtp.extensions);
- AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
- &video_config->rtp.extensions);
- AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri,
- &video_config->rtp.extensions);
+ 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));
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]);
- AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
- &video_config->rtp.extensions);
- AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
- &video_config->rtp.extensions);
+ video_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
+ video_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
}
void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
@@ -687,25 +666,6 @@
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 dbe8e07..fb26051 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -15,7 +15,6 @@
#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"
@@ -72,8 +71,6 @@
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);
@@ -219,13 +216,6 @@
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
new file mode 100644
index 0000000..4f33d25
--- /dev/null
+++ b/test/constants.cc
@@ -0,0 +1,27 @@
+/*
+ * 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
new file mode 100644
index 0000000..b1b87d6
--- /dev/null
+++ b/test/constants.h
@@ -0,0 +1,24 @@
+/*
+ * 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 830c2f1..7cff645 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -12,7 +12,6 @@
#include <map>
#include <memory>
-#include <utility>
#include <vector>
#include "api/test/simulated_network.h"
@@ -22,6 +21,7 @@
#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"
@@ -71,7 +71,14 @@
protected:
RtpRtcpObserver() : RtpRtcpObserver(0) {}
explicit RtpRtcpObserver(int event_timeout_ms)
- : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {}
+ : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
+ kTOffsetExtensionId);
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+ kAbsSendTimeExtensionId);
+ parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+ kTransportSequenceNumberExtensionId);
+ }
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 859626e..2f892f0 100644
--- a/test/scenario/video_stream.cc
+++ b/test/scenario/video_stream.cc
@@ -27,13 +27,8 @@
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 e7550a4..17bb985 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -26,11 +26,6 @@
#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:
@@ -47,8 +42,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
@@ -92,8 +87,8 @@
VideoEncoderConfig* encoder_config) override {
if (!send_side_bwe_) {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::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 eab9ef7..ed303ec 100644
--- a/video/end_to_end_tests/codec_tests.cc
+++ b/video/end_to_end_tests/codec_tests.cc
@@ -27,22 +27,11 @@
#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()) {
- RegisterRtpExtension(
- RtpExtension(RtpExtension::kColorSpaceUri, kColorSpaceExtensionId));
- RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
- kVideoRotationExtensionId));
- }
+ CodecEndToEndTest() : field_trial_(GetParam()) {}
private:
test::ScopedFieldTrials field_trial_;
@@ -235,10 +224,7 @@
class EndToEndTestH264 : public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
- EndToEndTestH264() : field_trial_(GetParam()) {
- RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
- kVideoRotationExtensionId));
- }
+ EndToEndTestH264() : field_trial_(GetParam()) {}
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 955f9cd..5b649d5 100644
--- a/video/end_to_end_tests/extended_reports_tests.cc
+++ b/video/end_to_end_tests/extended_reports_tests.cc
@@ -43,20 +43,8 @@
#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 {
- public:
- ExtendedReportsEndToEndTest() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
- }
-};
+class ExtendedReportsEndToEndTest : public test::CallTest {};
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 dbf0b2c..42215d3 100644
--- a/video/end_to_end_tests/fec_tests.cc
+++ b/video/end_to_end_tests/fec_tests.cc
@@ -22,21 +22,10 @@
#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() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
- RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
- kVideoRotationExtensionId));
- }
+ FecEndToEndTest() = default;
};
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 85ad7dd..98b3a4b 100644
--- a/video/end_to_end_tests/frame_encryption_tests.cc
+++ b/video/end_to_end_tests/frame_encryption_tests.cc
@@ -17,18 +17,10 @@
#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() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kGenericFrameDescriptorUri,
- kGenericDescriptorExtensionId));
- }
+ FrameEncryptionEndToEndTest() = default;
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 ef435b4..312924c 100644
--- a/video/end_to_end_tests/histogram_tests.cc
+++ b/video/end_to_end_tests/histogram_tests.cc
@@ -16,22 +16,8 @@
#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 dba3c3e..80c9ebd 100644
--- a/video/end_to_end_tests/probing_tests.cc
+++ b/video/end_to_end_tests/probing_tests.cc
@@ -15,21 +15,9 @@
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/gtest.h"
-
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 ProbingEndToEndTest : public test::CallTest {};
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 49c3fb4..7d31aed 100644
--- a/video/end_to_end_tests/retransmission_tests.cc
+++ b/video/end_to_end_tests/retransmission_tests.cc
@@ -21,18 +21,9 @@
#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() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
- kVideoRotationExtensionId));
- }
+ RetransmissionEndToEndTest() = default;
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 d89a024..bef15d2 100644
--- a/video/end_to_end_tests/rtp_rtcp_tests.cc
+++ b/video/end_to_end_tests/rtp_rtcp_tests.cc
@@ -18,11 +18,6 @@
#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:
@@ -535,7 +530,7 @@
flexfec_receive_config.transport_cc = true;
flexfec_receive_config.rtp_header_extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId);
+ test::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 bad04b4..53b181a 100644
--- a/video/end_to_end_tests/stats_tests.cc
+++ b/video/end_to_end_tests/stats_tests.cc
@@ -24,19 +24,7 @@
#include "test/rtcp_packet_parser.h"
namespace webrtc {
-namespace {
-enum : int { // The first valid value is 1.
- kVideoContentTypeExtensionId = 1,
-};
-} // namespace
-
-class StatsEndToEndTest : public test::CallTest {
- public:
- StatsEndToEndTest() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
- kVideoContentTypeExtensionId));
- }
-};
+class StatsEndToEndTest : public test::CallTest {};
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 b607bf6..9aacd9a 100644
--- a/video/end_to_end_tests/transport_feedback_tests.cc
+++ b/video/end_to_end_tests/transport_feedback_tests.cc
@@ -21,21 +21,12 @@
#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 {
- public:
- TransportFeedbackEndToEndTest() {
- RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
- }
-};
+class TransportFeedbackEndToEndTest : public test::CallTest {};
TEST_F(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
+ static const int kExtensionId = 5;
+
class RtpExtensionHeaderObserver : public test::DirectTransport {
public:
RtpExtensionHeaderObserver(
@@ -59,7 +50,7 @@
retransmit_observed_(false),
started_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
+ kExtensionId);
}
virtual ~RtpExtensionHeaderObserver() {}
@@ -183,9 +174,8 @@
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
// 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.
@@ -211,9 +201,8 @@
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,
- kTransportSequenceNumberExtensionId));
+ receive_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
receive_config->renderer = &fake_renderer_;
}
@@ -302,14 +291,14 @@
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*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_;
@@ -437,6 +426,7 @@
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:
@@ -445,7 +435,7 @@
video_observed_(false),
audio_observed_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
+ kExtensionId);
}
size_t GetNumVideoStreams() const override { return 1; }
@@ -455,9 +445,8 @@
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*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 4941827..ad3224d 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -47,14 +47,6 @@
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;
@@ -718,10 +710,10 @@
if (params_.call.send_side_bwe) {
video_send_configs_[video_idx].rtp.extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId);
+ test::kTransportSequenceNumberExtensionId);
} else {
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId);
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId);
}
if (params_.call.generic_descriptor) {
@@ -732,13 +724,13 @@
video_send_configs_[video_idx].rtp.extensions.emplace_back(
RtpExtension::kGenericFrameDescriptorUri,
- kGenericFrameDescriptorExtensionId);
+ test::kGenericDescriptorExtensionId);
}
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId);
+ RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId);
video_send_configs_[video_idx].rtp.extensions.emplace_back(
- RtpExtension::kVideoTimingUri, kVideoTimingExtensionId);
+ RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId);
video_encoder_configs_[video_idx].video_format.name =
params_.video[video_idx].codec;
@@ -886,10 +878,10 @@
if (params_.call.send_side_bwe) {
GetFlexFecConfig()->rtp_header_extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ test::kTransportSequenceNumberExtensionId));
} else {
- GetFlexFecConfig()->rtp_header_extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+ GetFlexFecConfig()->rtp_header_extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
}
@@ -927,10 +919,10 @@
if (params_.call.send_side_bwe) {
thumbnail_send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ test::kTransportSequenceNumberExtensionId));
} else {
- thumbnail_send_config.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+ thumbnail_send_config.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
VideoEncoderConfig thumbnail_encoder_config;
@@ -1362,7 +1354,7 @@
if (params_.call.send_side_bwe) {
audio_send_config.rtp.extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberExtensionId));
+ test::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 0beefec..057fd95 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -73,15 +73,6 @@
} // 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 {
@@ -93,12 +84,6 @@
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);
@@ -178,7 +163,7 @@
public:
AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
+ kRtpExtensionAbsoluteSendTime, test::kAbsSendTimeExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -208,8 +193,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
void PerformTest() override {
@@ -230,7 +215,7 @@
Clock::GetRealTimeClock(), kEncodeDelayMs);
}) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionTransmissionTimeOffset, kTimestampOffsetExtensionId));
+ kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
}
private:
@@ -254,7 +239,7 @@
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
+ RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
}
void PerformTest() override {
@@ -268,7 +253,7 @@
}
TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
- static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
+ static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
class TransportWideSequenceNumberObserver : public test::SendTest {
public:
TransportWideSequenceNumberObserver()
@@ -316,7 +301,7 @@
public:
VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoRotation, kVideoRotationExtensionId));
+ kRtpExtensionVideoRotation, test::kVideoRotationExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -337,7 +322,7 @@
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
+ RtpExtension::kVideoRotationUri, test::kVideoRotationExtensionId));
}
void OnFrameGeneratorCapturerCreated(
@@ -359,7 +344,7 @@
VideoContentTypeObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
- kRtpExtensionVideoContentType, kVideoContentTypeExtensionId));
+ kRtpExtensionVideoContentType, test::kVideoContentTypeExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -382,8 +367,9 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(RtpExtension(
- RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
+ send_config->rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kVideoContentTypeUri,
+ test::kVideoContentTypeExtensionId));
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
@@ -403,8 +389,8 @@
public:
VideoTimingObserver()
: SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
- EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(kRtpExtensionVideoTiming,
- kVideoTimingExtensionId));
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionVideoTiming, test::kVideoTimingExtensionId));
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
@@ -426,8 +412,8 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
- send_config->rtp.extensions.push_back(
- RtpExtension(RtpExtension::kVideoTimingUri, kVideoTimingExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId));
}
void PerformTest() override {
@@ -478,12 +464,7 @@
expect_ulpfec_(expect_ulpfec),
sent_media_(false),
sent_ulpfec_(false),
- header_extensions_enabled_(header_extensions_enabled) {
- parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
- kAbsSendTimeExtensionId);
- parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
- }
+ header_extensions_enabled_(header_extensions_enabled) {}
// Some of the test cases are expected to time out and thus we are using
// a shorter timeout window than the default here.
@@ -583,11 +564,12 @@
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, kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
(*receive_configs)[0].rtp.red_payload_type =
@@ -705,14 +687,7 @@
sent_media_(false),
sent_flexfec_(false),
header_extensions_enabled_(header_extensions_enabled),
- num_video_streams_(num_video_streams) {
- parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
- kAbsSendTimeExtensionId);
- parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
- kTimestampOffsetExtensionId);
- parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
- kTransportSequenceNumberExtensionId);
- }
+ num_video_streams_(num_video_streams) {}
size_t GetNumFlexfecStreams() const override { return 1; }
size_t GetNumVideoStreams() const override { return num_video_streams_; }
@@ -776,10 +751,10 @@
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::kTimestampOffsetUri, kTimestampOffsetExtensionId));
+ RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
} else {
send_config->rtp.extensions.clear();
}
@@ -1653,7 +1628,7 @@
TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
static const int kStartBitrateBps = 300000;
static const int kNewMaxBitrateBps = 1234567;
- static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
+ static const uint8_t kExtensionId = test::kTransportSequenceNumberExtensionId;
class ChangingNetworkRouteTest : public test::EndToEndTest {
public:
explicit ChangingNetworkRouteTest(
