webrtc_m130/call/bitrate_estimator_tests.cc

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/*
* Copyright (c) 2013 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.
*/
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 <cstddef>
#include <functional>
#include <list>
#include <memory>
#include <string>
#include "absl/strings/string_view.h"
#include "api/test/create_frame_generator.h"
#include "call/call.h"
#include "call/simulated_network.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread_annotations.h"
#include "test/call_test.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.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"
namespace webrtc {
namespace {
// Note: If you consider to re-use this class, think twice and instead consider
// writing tests that don't depend on the logging system.
class LogObserver {
public:
LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(absl::string_view expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
}
bool Wait() { return callback_.Wait(); }
private:
class Callback : public rtc::LogSink {
public:
void OnLogMessage(const std::string& message) override {
OnLogMessage(absl::string_view(message));
}
void OnLogMessage(absl::string_view message) override {
MutexLock lock(&mutex_);
// Ignore log lines that are due to missing AST extensions, these are
// logged when we switch back from AST to TOF until the wrapping bitrate
// estimator gives up on using AST.
if (message.find("BitrateEstimator") != absl::string_view::npos &&
message.find("packet is missing") == absl::string_view::npos) {
received_log_lines_.push_back(std::string(message));
}
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
std::string b = expected_log_lines_.front();
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
EXPECT_TRUE(a.find(b) != absl::string_view::npos) << a << " != " << b;
}
if (expected_log_lines_.empty()) {
if (num_popped > 0) {
done_.Set();
}
return;
}
}
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
bool Wait() {
return done_.Wait(test::VideoTestConstants::kDefaultTimeout);
}
void PushExpectedLogLine(absl::string_view expected_log_line) {
MutexLock lock(&mutex_);
expected_log_lines_.emplace_back(expected_log_line);
}
private:
typedef std::list<std::string> Strings;
Mutex mutex_;
Strings received_log_lines_ RTC_GUARDED_BY(mutex_);
Strings expected_log_lines_ RTC_GUARDED_BY(mutex_);
rtc::Event done_;
};
Callback callback_;
};
} // namespace
static const int kTOFExtensionId = 4;
static const int kASTExtensionId = 5;
class BitrateEstimatorTest : public test::CallTest {
public:
BitrateEstimatorTest() : receive_config_(nullptr) {}
virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
virtual void SetUp() {
SendTask(task_queue(), [this]() {
Reland "Delete PacketReceiver::DeliverPacket from all implementations" This reverts commit f2a083f262d86737893e774c696716742fcab3e3. Reason for revert: Test problem fixed in https://webrtc-review.googlesource.com/c/src/+/291333. Original change's description: > Revert "Delete PacketReceiver::DeliverPacket from all implementations" > > This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63. > > Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200 > > Original change's description: > > Delete PacketReceiver::DeliverPacket from all implementations > > > > And fix tests that still depend on extensions to be known by the receiver. > > > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > > > Bug: webrtc:7135,webrtc:14795 > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996 > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39184} > > Bug: webrtc:7135,webrtc:14795,b/266658815 > Change-Id: I9d03f4952938d176ffee110a707acadc1846457c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400 > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39189} Bug: webrtc:7135,webrtc:14795,b/266658815 Change-Id: Ia640f4342a1f42012ba5295003e17aef7613ad80 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291440 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39199}
2023-01-25 15:19:11 +01:00
RegisterRtpExtension(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
RegisterRtpExtension(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
CreateCalls();
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
CreateSendTransport(BuiltInNetworkBehaviorConfig(), /*observer=*/nullptr);
CreateReceiveTransport(BuiltInNetworkBehaviorConfig(),
/*observer=*/nullptr);
VideoSendStream::Config video_send_config(send_transport_.get());
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_send_config.rtp.ssrcs.push_back(
test::VideoTestConstants::kVideoSendSsrcs[0]);
video_send_config.encoder_settings.encoder_factory =
&fake_encoder_factory_;
video_send_config.encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
video_send_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_send_config.rtp.payload_type =
test::VideoTestConstants::kFakeVideoSendPayloadType;
SetVideoSendConfig(video_send_config);
VideoEncoderConfig video_encoder_config;
test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
SetVideoEncoderConfig(video_encoder_config);
receive_config_ =
VideoReceiveStreamInterface::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[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
receive_config_.rtp.local_ssrc =
test::VideoTestConstants::kReceiverLocalVideoSsrc;
});
}
virtual void TearDown() {
SendTask(task_queue(), [this]() {
for (auto* stream : streams_) {
stream->StopSending();
delete stream;
}
streams_.clear();
DestroyCalls();
});
}
protected:
friend class Stream;
class Stream {
public:
explicit Stream(BitrateEstimatorTest* test)
: test_(test),
is_sending_receiving_(false),
send_stream_(nullptr),
frame_generator_capturer_(),
decoder_factory_(
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
[]() { return std::make_unique<test::FakeDecoder>(); }) {
test_->GetVideoSendConfig()->rtp.ssrcs[0]++;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
test_->GetVideoSendConfig()->Copy(),
test_->GetVideoEncoderConfig()->Copy());
RTC_DCHECK_EQ(1, test_->GetVideoEncoderConfig()->number_of_streams);
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>(
test->clock_,
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
test::CreateSquareFrameGenerator(
test::VideoTestConstants::kDefaultWidth,
test::VideoTestConstants::kDefaultHeight, absl::nullopt,
absl::nullopt),
test::VideoTestConstants::kDefaultFramerate,
*test->task_queue_factory_);
frame_generator_capturer_->Init();
frame_generator_capturer_->Start();
send_stream_->SetSource(frame_generator_capturer_.get(),
DegradationPreference::MAINTAIN_FRAMERATE);
send_stream_->Start();
VideoReceiveStreamInterface::Decoder decoder;
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
test_->receive_config_.decoder_factory = &decoder_factory_;
decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
decoder.video_format =
SdpVideoFormat(test_->GetVideoSendConfig()->rtp.payload_name);
test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
test_->GetVideoSendConfig()->rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
test_->receive_config_.renderer = &test->fake_renderer_;
video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_.Copy());
video_receive_stream_->Start();
is_sending_receiving_ = true;
}
~Stream() {
EXPECT_FALSE(is_sending_receiving_);
test_->sender_call_->DestroyVideoSendStream(send_stream_);
frame_generator_capturer_.reset(nullptr);
send_stream_ = nullptr;
if (video_receive_stream_) {
test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
video_receive_stream_ = nullptr;
}
}
void StopSending() {
if (is_sending_receiving_) {
send_stream_->Stop();
if (video_receive_stream_) {
video_receive_stream_->Stop();
}
is_sending_receiving_ = false;
}
}
private:
BitrateEstimatorTest* test_;
bool is_sending_receiving_;
VideoSendStream* send_stream_;
VideoReceiveStreamInterface* video_receive_stream_;
std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
test::FunctionVideoDecoderFactory decoder_factory_;
};
LogObserver receiver_log_;
VideoReceiveStreamInterface::Config receive_config_;
std::vector<Stream*> streams_;
};
static const char* kAbsSendTimeLog =
"RemoteBitrateEstimatorAbsSendTime: Instantiating.";
static const char* kSingleStreamLog =
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
}
// This test is flaky. See webrtc:5790.
TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
streams_.push_back(new Stream(this));
});
EXPECT_TRUE(receiver_log_.Wait());
SendTask(task_queue(), [this]() {
GetVideoSendConfig()->rtp.extensions[0] =
RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
streams_.push_back(new Stream(this));
streams_[0]->StopSending();
streams_[1]->StopSending();
});
EXPECT_TRUE(receiver_log_.Wait());
}
} // namespace webrtc