webrtc_m130/test/call_test.cc

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/*
* Copyright (c) 2014 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/call_test.h"
#include <algorithm>
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
#include <memory>
#include "api/audio/audio_device.h"
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/create_frame_generator.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/fake_network_pipe.h"
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
#include "call/packet_receiver.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/fake_encoder.h"
#include "test/network/simulated_network.h"
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
#include "test/rtp_rtcp_observer.h"
#include "test/testsupport/file_utils.h"
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
#include "test/video_test_constants.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
namespace test {
CallTest::CallTest()
: env_(CreateEnvironment(&field_trials_)),
send_env_(env_),
recv_env_(env_),
audio_send_config_(/*send_transport=*/nullptr),
audio_send_stream_(nullptr),
frame_generator_capturer_(nullptr),
fake_encoder_factory_(
[this](const Environment& env, const SdpVideoFormat& format) {
std::unique_ptr<FakeEncoder> fake_encoder;
if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
fake_encoder = std::make_unique<FakeVp8Encoder>(env);
} else {
fake_encoder = std::make_unique<FakeEncoder>(env);
}
fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
return fake_encoder;
}),
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
fake_decoder_factory_([]() { return std::make_unique<FakeDecoder>(); }),
bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
num_video_streams_(1),
num_audio_streams_(0),
num_flexfec_streams_(0),
audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
task_queue_(env_.task_queue_factory().CreateTaskQueue(
"CallTestTaskQueue",
TaskQueueFactory::Priority::NORMAL)) {}
CallTest::~CallTest() = default;
void CallTest::SetSendEventLog(std::unique_ptr<RtcEventLog> event_log) {
EnvironmentFactory f(env_);
f.Set(std::move(event_log));
send_env_ = f.Create();
}
void CallTest::SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log) {
EnvironmentFactory f(env_);
f.Set(std::move(event_log));
recv_env_ = f.Create();
}
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}
2019-02-18 23:45:57 +01:00
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 << ").";
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}
2019-02-18 23:45:57 +01:00
}
}
rtp_extensions_.push_back(extension);
}
void CallTest::RunBaseTest(BaseTest* test) {
SendTask(task_queue(), [this, test]() {
num_video_streams_ = test->GetNumVideoStreams();
num_audio_streams_ = test->GetNumAudioStreams();
num_flexfec_streams_ = test->GetNumFlexfecStreams();
RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
CallConfig send_config = SendCallConfig();
CallConfig recv_config = RecvCallConfig();
test->ModifySenderBitrateConfig(&send_config.bitrate_config);
if (num_audio_streams_ > 0) {
CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
fake_recv_audio_device_.get());
apm_send_ = BuiltinAudioProcessingBuilder().Build(send_config.env);
apm_recv_ = BuiltinAudioProcessingBuilder().Build(recv_config.env);
EXPECT_EQ(0, fake_send_audio_device_->Init());
EXPECT_EQ(0, fake_recv_audio_device_->Init());
AudioState::Config audio_state_config;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_send_;
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
audio_state_config.audio_device_module = fake_send_audio_device_;
send_config.audio_state = AudioState::Create(audio_state_config);
fake_send_audio_device_->RegisterAudioCallback(
send_config.audio_state->audio_transport());
}
CreateSenderCall(std::move(send_config));
if (test->ShouldCreateReceivers()) {
test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
if (num_audio_streams_ > 0) {
AudioState::Config audio_state_config;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
audio_state_config.audio_processing = apm_recv_;
Remove voe::TransmitMixer TransmitMixer's functionality is moved into the AudioTransportProxy owned by AudioState. This removes the need for an AudioTransport implementation in VoEBaseImpl, which means that the proxy is no longer a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl. In the short term, AudioState needs to know which AudioDeviceModule is used, so it is added in AudioState::Config. AudioTransportImpl needs to know which AudioSendStream:s are currently enabled to send, so AudioState maintains a map of them, which is reduced into a simple vector for AudioTransportImpl. To encode and transmit audio, AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame) is introduced, which is used in both the Chromium and standalone use cases. This removes the need for two different instances of voe::Channel::ProcessAndEncodeAudio(), so there is now only one, taking an AudioFrame as argument. Callers need to allocate their own AudioFrame:s, which is wasteful but not a regression since this was already happening in the voe::Channel functions. Most of the logic changed resides in AudioTransportImpl::RecordedDataIsAvailable(), where two strange things were found: 1. The clock drift parameter was ineffective since apm->echo_cancellation()->enable_drift_compensation(false) is called during initialization. 2. The output parameter 'new_mic_volume' was never set - instead it was returned as a result, causing the ADM to never update the analog mic gain (https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100). Besides this, tests are updated, and some dead code is removed which was found in the process. Bug: webrtc:4690, webrtc:8591 Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2 Reviewed-on: https://webrtc-review.googlesource.com/26681 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
audio_state_config.audio_device_module = fake_recv_audio_device_;
recv_config.audio_state = AudioState::Create(audio_state_config);
fake_recv_audio_device_->RegisterAudioCallback(
recv_config.audio_state->audio_transport());
}
CreateReceiverCall(std::move(recv_config));
}
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
CreateReceiveTransport(test->GetReceiveTransportConfig(), test);
CreateSendTransport(test->GetSendTransportConfig(), test);
test->OnTransportCreated(send_transport_.get(), send_simulated_network_,
receive_transport_.get(),
receive_simulated_network_);
if (test->ShouldCreateReceivers()) {
if (num_video_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
if (num_audio_streams_ > 0)
receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
} else {
// Sender-only call delivers to itself.
send_transport_->SetReceiver(sender_call_->Receiver());
receive_transport_->SetReceiver(nullptr);
}
CreateSendConfig(num_video_streams_, num_audio_streams_,
num_flexfec_streams_, send_transport_.get());
if (test->ShouldCreateReceivers()) {
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
CreateMatchingReceiveConfigs();
}
if (num_video_streams_ > 0) {
test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
GetVideoEncoderConfig());
}
if (num_audio_streams_ > 0) {
test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
}
if (num_flexfec_streams_ > 0) {
CreateFlexfecStreams();
test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
}
if (num_video_streams_ > 0) {
CreateVideoStreams();
test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
}
if (num_audio_streams_ > 0) {
CreateAudioStreams();
test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
}
if (num_video_streams_ > 0) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
int width = VideoTestConstants::kDefaultWidth;
int height = VideoTestConstants::kDefaultHeight;
int frame_rate = VideoTestConstants::kDefaultFramerate;
test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
test->ModifyVideoDegradationPreference(&degradation_preference_);
CreateFrameGeneratorCapturer(frame_rate, width, height);
test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
}
Start();
});
test->PerformTest();
SendTask(task_queue(), [this, test]() {
Stop();
test->OnStreamsStopped();
DestroyStreams();
send_transport_.reset();
receive_transport_.reset();
frame_generator_capturer_ = nullptr;
DestroyCalls();
fake_send_audio_device_ = nullptr;
fake_recv_audio_device_ = nullptr;
});
}
CallConfig CallTest::SendCallConfig() const {
CallConfig sender_config(send_env_);
sender_config.network_state_predictor_factory =
network_state_predictor_factory_.get();
sender_config.network_controller_factory = network_controller_factory_.get();
return sender_config;
}
CallConfig CallTest::RecvCallConfig() const {
return CallConfig(recv_env_);
}
void CallTest::CreateCalls() {
CreateCalls(SendCallConfig(), RecvCallConfig());
}
void CallTest::CreateCalls(CallConfig sender_config,
CallConfig receiver_config) {
CreateSenderCall(std::move(sender_config));
CreateReceiverCall(std::move(receiver_config));
}
void CallTest::CreateSenderCall() {
CreateSenderCall(SendCallConfig());
}
void CallTest::CreateSenderCall(CallConfig config) {
sender_call_ = Call::Create(std::move(config));
}
void CallTest::CreateReceiverCall(CallConfig config) {
receiver_call_ = Call::Create(std::move(config));
}
void CallTest::DestroyCalls() {
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
send_transport_.reset();
receive_transport_.reset();
sender_call_.reset();
receiver_call_.reset();
}
void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
size_t num_video_streams,
size_t num_used_ssrcs,
Transport* send_transport) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
RTC_DCHECK_LE(num_video_streams + num_used_ssrcs,
VideoTestConstants::kNumSsrcs);
*video_config = VideoSendStream::Config(send_transport);
video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
video_config->encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
Reland "Reland "Move rtp-specific config out of EncoderSettings."" This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e. Reason for revert: Intend to investigate and fix perf problems. Original change's description: > Revert "Reland "Move rtp-specific config out of EncoderSettings."" > > This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266. > > Reason for revert: Regression in ramp up perf tests. > > Original change's description: > > Reland "Move rtp-specific config out of EncoderSettings." > > > > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c > > > > Original change's description: > > > Move rtp-specific config out of EncoderSettings. > > > > > > In VideoSendStream::Config, move payload_name and payload_type from > > > EncoderSettings to Rtp. > > > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > > and should perhaps be renamed in a follow up cl. It's no longer > > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > > > The latter then needs a different way to know the codec type, > > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > > > Bug: webrtc:8830 > > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22532} > > > > Bug: webrtc:8830 > > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > > Reviewed-on: https://webrtc-review.googlesource.com/63721 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22595} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org > > Bug: webrtc:8830,chromium:827080 > Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef > Reviewed-on: https://webrtc-review.googlesource.com/65520 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22677} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8830, chromium:827080 Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f Reviewed-on: https://webrtc-review.googlesource.com/66862 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22751}
2018-04-05 15:36:51 +02:00
video_config->rtp.payload_name = "FAKE";
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
video_config->rtp.payload_type =
VideoTestConstants::kFakeVideoSendPayloadType;
video_config->rtp.extmap_allow_mixed = true;
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}
2019-02-18 23:45:57 +01:00
AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kAbsSendTimeUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kTimestampOffsetUri,
&video_config->rtp.extensions);
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}
2019-02-18 23:45:57 +01:00
AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri00,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kDependencyDescriptorUri,
&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)
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
video_config->rtp.ssrcs.push_back(
VideoTestConstants::kVideoSendSsrcs[num_used_ssrcs + i]);
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}
2019-02-18 23:45:57 +01:00
AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
&video_config->rtp.extensions);
AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
&video_config->rtp.extensions);
}
void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport) {
RTC_DCHECK_LE(num_audio_streams, 1);
RTC_DCHECK_LE(num_flexfec_streams, 1);
if (num_audio_streams > 0) {
AudioSendStream::Config audio_send_config(send_transport);
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
audio_send_config.rtp.ssrc = VideoTestConstants::kAudioSendSsrc;
AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
&audio_send_config.rtp.extensions);
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
VideoTestConstants::kAudioSendPayloadType,
{"opus", 48000, 2, {{"stereo", "1"}}});
audio_send_config.min_bitrate_bps = 6000;
audio_send_config.max_bitrate_bps = 60000;
audio_send_config.encoder_factory = audio_encoder_factory_;
SetAudioConfig(audio_send_config);
}
// TODO(brandtr): Update this when we support multistream protection.
if (num_flexfec_streams > 0) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
SetSendFecConfig({VideoTestConstants::kVideoSendSsrcs[0]});
}
}
void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
audio_send_config_ = config;
}
void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
GetVideoSendConfig()->rtp.flexfec.payload_type =
VideoTestConstants::kFlexfecPayloadType;
GetVideoSendConfig()->rtp.flexfec.ssrc = VideoTestConstants::kFlexfecSendSsrc;
GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
}
void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
send_config->rtp.ulpfec.red_payload_type =
VideoTestConstants::kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type =
VideoTestConstants::kUlpfecPayloadType;
send_config->rtp.ulpfec.red_rtx_payload_type =
VideoTestConstants::kRtxRedPayloadType;
}
void CallTest::SetReceiveUlpFecConfig(
VideoReceiveStreamInterface::Config* receive_config) {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
receive_config->rtp.red_payload_type = VideoTestConstants::kRedPayloadType;
receive_config->rtp.ulpfec_payload_type =
VideoTestConstants::kUlpfecPayloadType;
receive_config->rtp
.rtx_associated_payload_types[VideoTestConstants::kRtxRedPayloadType] =
VideoTestConstants::kRedPayloadType;
}
void CallTest::CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport) {
if (num_video_streams > 0) {
video_send_configs_.clear();
video_send_configs_.emplace_back(nullptr);
CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
send_transport);
}
CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
send_transport);
}
void CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport) {
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
&fake_decoder_factory_, std::nullopt, false,
0);
}
void CallTest::CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
AddMatchingVideoReceiveConfigs(
&video_receive_configs_, video_send_config, rtcp_send_transport,
decoder_factory, decode_sub_stream, receiver_reference_time_report,
rtp_history_ms);
}
void CallTest::AddMatchingVideoReceiveConfigs(
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
VideoDecoderFactory* decoder_factory,
std::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms) {
RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
VideoReceiveStreamInterface::Config default_config(rtcp_send_transport);
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
default_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc;
default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
// Enable RTT calculation so NTP time estimator will work.
default_config.rtp.rtcp_xr.receiver_reference_time_report =
receiver_reference_time_report;
default_config.renderer = &fake_renderer_;
for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
VideoReceiveStreamInterface::Config video_recv_config(
default_config.Copy());
video_recv_config.decoders.clear();
if (!video_send_config.rtp.rtx.ssrcs.empty()) {
video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
video_recv_config.rtp.rtx_associated_payload_types
[VideoTestConstants::kSendRtxPayloadType] =
video_send_config.rtp.payload_type;
}
video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
VideoReceiveStreamInterface::Decoder decoder;
decoder.payload_type = video_send_config.rtp.payload_type;
decoder.video_format = SdpVideoFormat(video_send_config.rtp.payload_name);
// Force fake decoders on non-selected simulcast streams.
if (!decode_sub_stream || i == *decode_sub_stream) {
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8. Reason for revert: Downstream fix landed. TBR=mflodman@webrtc.org Original change's description: > Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." > > This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759. > > Reason for revert: Break downstream stuff. > > Original change's description: > > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory. > > > > Bug: webrtc:9106 > > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31834} > > TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org > > Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9106 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31835} TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9106 Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31866}
2020-08-03 15:55:10 +00:00
video_recv_config.decoder_factory = decoder_factory;
} else {
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." This reverts commit a4f23ad0ce4382e3a11bc6a8c1f9f6183e722fd8. Reason for revert: Downstream fix landed. TBR=mflodman@webrtc.org Original change's description: > Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." > > This reverts commit acb9d8365a5f9eb1e2a9e9902690d62dab1e5759. > > Reason for revert: Break downstream stuff. > > Original change's description: > > Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory. > > > > Bug: webrtc:9106 > > Change-Id: I85712f3ab6a734d3fad7819491d3b8e3388b47e7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180342 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31834} > > TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org > > Change-Id: I6cfdb85a154a78135839f84edf5f69673d5ab715 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9106 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180807 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31835} TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9106 Change-Id: I03b3e68532107bec37bcc6e47a5489c84fe91ef9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180808 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31866}
2020-08-03 15:55:10 +00:00
video_recv_config.decoder_factory = &fake_decoder_factory_;
}
video_recv_config.decoders.push_back(decoder);
receive_configs->emplace_back(std::move(video_recv_config));
}
}
void CallTest::CreateMatchingAudioAndFecConfigs(
Transport* rtcp_send_transport) {
RTC_DCHECK_GE(1, num_audio_streams_);
if (num_audio_streams_ == 1) {
CreateMatchingAudioConfigs(rtcp_send_transport, "");
}
// TODO(brandtr): Update this when we support multistream protection.
RTC_DCHECK(num_flexfec_streams_ <= 1);
if (num_flexfec_streams_ == 1) {
CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
}
}
void CallTest::CreateMatchingAudioConfigs(Transport* transport,
std::string sync_group) {
audio_receive_configs_.push_back(CreateMatchingAudioConfig(
audio_send_config_, audio_decoder_factory_, transport, sync_group));
}
AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig(
const AudioSendStream::Config& send_config,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
Transport* transport,
std::string sync_group) {
AudioReceiveStreamInterface::Config audio_config;
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
audio_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = transport;
audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
audio_config.decoder_factory = audio_decoder_factory;
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
audio_config.decoder_map = {
{VideoTestConstants::kAudioSendPayloadType, {"opus", 48000, 2}}};
audio_config.sync_group = sync_group;
return audio_config;
}
void CallTest::CreateMatchingFecConfig(
Transport* transport,
const VideoSendStream::Config& send_config) {
FlexfecReceiveStream::Config config(transport);
config.payload_type = send_config.rtp.flexfec.payload_type;
config.rtp.remote_ssrc = send_config.rtp.flexfec.ssrc;
config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc;
if (!video_receive_configs_.empty()) {
video_receive_configs_[0].rtp.protected_by_flexfec = true;
video_receive_configs_[0].rtp.packet_sink_ = this;
}
flexfec_receive_configs_.push_back(config);
}
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
video_receive_configs_.clear();
for (VideoSendStream::Config& video_send_config : video_send_configs_) {
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport);
}
CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
}
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
float speed,
int framerate,
int width,
int height) {
video_sources_.clear();
auto frame_generator_capturer =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<test::FrameGeneratorCapturer>(
clock,
test::CreateSquareFrameGenerator(width, height, std::nullopt,
std::nullopt),
framerate * speed, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
ConnectVideoSourcesToStreams();
}
void CallTest::CreateFrameGeneratorCapturer(int framerate,
int width,
int height) {
video_sources_.clear();
auto frame_generator_capturer =
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
std::make_unique<test::FrameGeneratorCapturer>(
&env_.clock(),
test::CreateSquareFrameGenerator(width, height, std::nullopt,
std::nullopt),
framerate, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
ConnectVideoSourcesToStreams();
}
void CallTest::CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" This reverts commit fd5166c305068772d00ad7edf50151bba215400b. Reason for revert: Stop using CreateTestAudioDeviceModule in downstream Original change's description: > Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" > > This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2. > > Reason for revert: Breaks downstream importer. > > Original change's description: > > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory > > > > Bug: webrtc:10284 > > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28227} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10284 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28228} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10284 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 11:49:17 +00:00
fake_send_audio_device_ = TestAudioDeviceModule::Create(
&env_.task_queue_factory(), std::move(capturer), nullptr, 1.f);
Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" This reverts commit fd5166c305068772d00ad7edf50151bba215400b. Reason for revert: Stop using CreateTestAudioDeviceModule in downstream Original change's description: > Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" > > This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2. > > Reason for revert: Breaks downstream importer. > > Original change's description: > > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory > > > > Bug: webrtc:10284 > > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28227} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10284 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28228} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10284 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 11:49:17 +00:00
fake_recv_audio_device_ = TestAudioDeviceModule::Create(
&env_.task_queue_factory(), nullptr, std::move(renderer), 1.f);
}
void CallTest::CreateVideoStreams() {
RTC_DCHECK(video_receive_streams_.empty());
CreateVideoSendStreams();
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
video_receive_configs_[i].Copy()));
}
}
void CallTest::CreateVideoSendStreams() {
RTC_DCHECK(video_send_streams_.empty());
// We currently only support testing external fec controllers with a single
// VideoSendStream.
if (fec_controller_factory_.get()) {
RTC_DCHECK_LE(video_send_configs_.size(), 1);
}
// TODO(http://crbug/818127):
// Remove this workaround when ALR is not screenshare-specific.
std::list<size_t> streams_creation_order;
for (size_t i = 0; i < video_send_configs_.size(); ++i) {
// If dual streams are created, add the screenshare stream last.
if (video_encoder_configs_[i].content_type ==
VideoEncoderConfig::ContentType::kScreen) {
streams_creation_order.push_back(i);
} else {
streams_creation_order.push_front(i);
}
}
video_send_streams_.resize(video_send_configs_.size(), nullptr);
for (size_t i : streams_creation_order) {
if (fec_controller_factory_.get()) {
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
fec_controller_factory_->CreateFecController(send_env_));
} else {
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
}
}
}
void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) {
RTC_DCHECK(video_send_streams_.empty());
video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
GetVideoSendConfig()->Copy(), encoder_config.Copy()));
}
void CallTest::CreateAudioStreams() {
RTC_DCHECK(audio_send_stream_ == nullptr);
RTC_DCHECK(audio_receive_streams_.empty());
audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
audio_receive_streams_.push_back(
receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
}
}
void CallTest::CreateFlexfecStreams() {
for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
flexfec_receive_streams_.push_back(
receiver_call_->CreateFlexfecReceiveStream(
flexfec_receive_configs_[i]));
}
}
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
void CallTest::CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer) {
PacketReceiver* receiver =
receiver_call_ ? receiver_call_->Receiver() : nullptr;
auto network = std::make_unique<SimulatedNetwork>(config);
send_simulated_network_ = network.get();
send_transport_ = std::make_unique<PacketTransport>(
task_queue(), sender_call_.get(), observer,
test::PacketTransport::kSender, payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network), receiver),
rtp_extensions_, rtp_extensions_);
}
void CallTest::CreateReceiveTransport(
const BuiltInNetworkBehaviorConfig& config,
RtpRtcpObserver* observer) {
auto network = std::make_unique<SimulatedNetwork>(config);
receive_simulated_network_ = network.get();
receive_transport_ = std::make_unique<PacketTransport>(
task_queue(), nullptr, observer, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network),
sender_call_->Receiver()),
rtp_extensions_, rtp_extensions_);
}
void CallTest::ConnectVideoSourcesToStreams() {
for (size_t i = 0; i < video_sources_.size(); ++i)
video_send_streams_[i]->SetSource(video_sources_[i].get(),
degradation_preference_);
}
void CallTest::Start() {
StartVideoStreams();
if (audio_send_stream_) {
audio_send_stream_->Start();
}
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Start();
}
void CallTest::StartVideoSources() {
for (size_t i = 0; i < video_sources_.size(); ++i) {
video_sources_[i]->Start();
}
}
void CallTest::StartVideoStreams() {
StartVideoSources();
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880. Reason for revert: Perf regression not affecting open source. Original change's description: > Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" > > This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc. > > Reason for revert: Tentative revert due to possible perf regression. b/260123362 > > Original change's description: > > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream > > > > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. > > Therefore this cl: > > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. > > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. > > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. > > > > Bug: none > > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38698} > > Bug: none > Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38725} Bug: b/260400659 Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38794}
2022-12-02 08:09:37 +00:00
for (size_t i = 0; i < video_send_streams_.size(); ++i) {
video_send_streams_[i]->Start();
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880. Reason for revert: Perf regression not affecting open source. Original change's description: > Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" > > This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc. > > Reason for revert: Tentative revert due to possible perf regression. b/260123362 > > Original change's description: > > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream > > > > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. > > Therefore this cl: > > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. > > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. > > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. > > > > Bug: none > > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38698} > > Bug: none > Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38725} Bug: b/260400659 Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38794}
2022-12-02 08:09:37 +00:00
}
for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
video_recv_stream->Start();
}
void CallTest::Stop() {
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
audio_recv_stream->Stop();
if (audio_send_stream_) {
audio_send_stream_->Stop();
}
StopVideoStreams();
}
void CallTest::StopVideoStreams() {
for (VideoSendStream* video_send_stream : video_send_streams_)
video_send_stream->Stop();
for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
video_recv_stream->Stop();
}
void CallTest::DestroyStreams() {
if (audio_send_stream_)
sender_call_->DestroyAudioSendStream(audio_send_stream_);
audio_send_stream_ = nullptr;
for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
DestroyVideoSendStreams();
for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
video_receive_streams_.clear();
video_sources_.clear();
}
void CallTest::DestroyVideoSendStreams() {
for (VideoSendStream* video_send_stream : video_send_streams_)
sender_call_->DestroyVideoSendStream(video_send_stream);
video_send_streams_.clear();
}
void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
frame_generator_capturer_->SetFakeRotation(rotation);
}
void CallTest::SetVideoDegradation(DegradationPreference preference) {
GetVideoSendStream()->SetSource(frame_generator_capturer_, preference);
}
VideoSendStream::Config* CallTest::GetVideoSendConfig() {
return &video_send_configs_[0];
}
void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) {
video_send_configs_.clear();
video_send_configs_.push_back(config.Copy());
}
VideoEncoderConfig* CallTest::GetVideoEncoderConfig() {
return &video_encoder_configs_[0];
}
void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) {
video_encoder_configs_.clear();
video_encoder_configs_.push_back(config.Copy());
}
VideoSendStream* CallTest::GetVideoSendStream() {
return video_send_streams_[0];
}
FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() {
return &flexfec_receive_configs_[0];
}
void CallTest::OnRtpPacket(const RtpPacketReceived& packet) {
// All FlexFEC streams protect all of the video streams.
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
flexfec_recv_stream->OnRtpPacket(packet);
}
std::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
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}
2019-02-18 23:45:57 +01:00
const std::string& uri) const {
for (const auto& extension : rtp_extensions_) {
if (extension.uri == uri) {
return extension;
}
}
return std::nullopt;
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}
2019-02-18 23:45:57 +01:00
}
void CallTest::AddRtpExtensionByUri(
const std::string& uri,
std::vector<RtpExtension>* extensions) const {
const std::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
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}
2019-02-18 23:45:57 +01:00
if (extension) {
extensions->push_back(*extension);
}
}
const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
Reland "Remove dependency of video_replay on TestADM." This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a. Reason for revert: reland with fix Original change's description: > Revert "Remove dependency of video_replay on TestADM." > > This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67. > > Reason for revert: breaking CallPerfTest > https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview > > Original change's description: > > Remove dependency of video_replay on TestADM. > > > > This should remove requirement to build TestADM in chromium build. > > > > Bug: b/272350185, webrtc:15081 > > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380 > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39934} > > Bug: b/272350185, webrtc:15081 > Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39939} Bug: b/272350185, webrtc:15081 Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Auto-Submit: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:56:49 +02:00
{VideoTestConstants::kVideoSendPayloadType, MediaType::VIDEO},
{VideoTestConstants::kFakeVideoSendPayloadType, MediaType::VIDEO},
{VideoTestConstants::kSendRtxPayloadType, MediaType::VIDEO},
{VideoTestConstants::kPayloadTypeVP8, MediaType::VIDEO},
{VideoTestConstants::kPayloadTypeVP9, MediaType::VIDEO},
{VideoTestConstants::kPayloadTypeH264, MediaType::VIDEO},
{VideoTestConstants::kPayloadTypeGeneric, MediaType::VIDEO},
{VideoTestConstants::kRedPayloadType, MediaType::VIDEO},
{VideoTestConstants::kRtxRedPayloadType, MediaType::VIDEO},
{VideoTestConstants::kUlpfecPayloadType, MediaType::VIDEO},
{VideoTestConstants::kFlexfecPayloadType, MediaType::VIDEO},
{VideoTestConstants::kAudioSendPayloadType, MediaType::AUDIO}};
BaseTest::BaseTest() {}
BaseTest::BaseTest(TimeDelta timeout) : RtpRtcpObserver(timeout) {}
BaseTest::~BaseTest() {}
std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() {
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
}
std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
return TestAudioDeviceModule::CreateDiscardRenderer(48000);
}
void BaseTest::OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device) {
}
void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
}
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
void BaseTest::OnTransportCreated(PacketTransport* to_receiver,
SimulatedNetworkInterface* sender_network,
PacketTransport* to_sender,
SimulatedNetworkInterface* receiver_network) {
}
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-19 15:45:58 +00:00
BuiltInNetworkBehaviorConfig BaseTest::GetSendTransportConfig() const {
return BuiltInNetworkBehaviorConfig();
}
BuiltInNetworkBehaviorConfig BaseTest::GetReceiveTransportConfig() const {
return BuiltInNetworkBehaviorConfig();
}
size_t BaseTest::GetNumVideoStreams() const {
return 1;
}
size_t BaseTest::GetNumAudioStreams() const {
return 0;
}
size_t BaseTest::GetNumFlexfecStreams() const {
return 0;
}
void BaseTest::ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) {}
void BaseTest::ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate) {}
void BaseTest::ModifyVideoDegradationPreference(
DegradationPreference* degradation_preference) {}
void BaseTest::OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>& receive_streams) {}
void BaseTest::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {}
void BaseTest::OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>& receive_streams) {}
void BaseTest::ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
void BaseTest::OnFlexfecStreamsCreated(
const std::vector<FlexfecReceiveStream*>& receive_streams) {}
void BaseTest::OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer) {}
void BaseTest::OnStreamsStopped() {}
SendTest::SendTest(TimeDelta timeout) : BaseTest(timeout) {}
bool SendTest::ShouldCreateReceivers() const {
return false;
}
EndToEndTest::EndToEndTest() {}
EndToEndTest::EndToEndTest(TimeDelta timeout) : BaseTest(timeout) {}
bool EndToEndTest::ShouldCreateReceivers() const {
return true;
}
} // namespace test
} // namespace webrtc