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Revert "Introduce ability to test echo in PC level test framework"

Browse source 
This reverts commit 77acb015b6.

Reason for revert: Downstream tests are failing.

Original change's description:
> Introduce ability to test echo in PC level test framework
> 
> Bug: webrtc:10138
> Change-Id: Ie638eaec5a46e37dc0eb52e9432fdebd0e4a1c4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147866
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28892}

TBR=mbonadei@webrtc.org,saza@webrtc.org,kwiberg@webrtc.org,titovartem@webrtc.org

Change-Id: Idc87c1cb679712d701d30902bcae4e2c698cf1cd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149804
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28896}
This commit is contained in:
Sami Kalliomäki 2019-08-19 11:45:56 +00:00 committed by Commit Bot
parent 93f518917f
commit 5870503d5e
9 changed files with 40 additions and 305 deletions
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11
api/test/peerconnection_quality_test_fixture.h
View file

@ -279,13 +279,6 @@ class PeerConnectionE2EQualityTestFixture {
PeerConnectionInterface::BitrateParameters bitrate_params) = 0;
};
// Contains configuration for echo emulator.
struct EchoEmulationConfig {
// Delay which represents the echo path delay, i.e. how soon rendered signal
// should reach capturer.
TimeDelta echo_delay = TimeDelta::ms(50);
};
// Contains parameters, that describe how long framework should run quality
// test.
struct RunParams {
@ -321,10 +314,6 @@ class PeerConnectionE2EQualityTestFixture {
// If true will set conference mode in SDP media section for all video
// tracks for all peers.
bool use_conference_mode = false;
// If specified echo emulation will be done, by mixing the render audio into
// the capture signal. In such case input signal will be reduced by half to
// avoid saturation or compression in the echo path simulation.
absl::optional<EchoEmulationConfig> echo_emulation_config;
};
// Represent an entity that will report quality metrics after test.

10
rtc_base/swap_queue.h
View file

@ -200,16 +200,6 @@ class SwapQueue {
return true;
}
// Returns the current number of elements in the queue. Since elements may be
// concurrently added to the queue, the caller must treat this as a lower
// bound, not an exact count.
// May only be called by the consumer.
size_t SizeAtLeast() const {
// Acquire memory ordering ensures that we wait for the producer to finish
// inserting any element in progress.
return std::atomic_load_explicit(&num_elements_, std::memory_order_acquire);
}
private:
// Verify that the queue slots complies with the ItemVerifier test. This
// function is not thread-safe and can only be used in the constructors.

15
test/pc/e2e/BUILD.gn
View file

@ -203,20 +203,6 @@ if (rtc_include_tests) {
]
}
rtc_source_set("echo_emulation") {
visibility = [ "*" ]
testonly = true
sources = [
"echo/echo_emulation.cc",
"echo/echo_emulation.h",
]
deps = [
"../../../api:peer_connection_quality_test_fixture_api",
"../../../modules/audio_device:audio_device_impl",
"../../../rtc_base:rtc_base_approved",
]
}
rtc_source_set("test_peer") {
visibility = [ "*" ]
testonly = true
@ -225,7 +211,6 @@ if (rtc_include_tests) {
"test_peer.h",
]
deps = [
":echo_emulation",
":peer_connection_quality_test_params",
":video_quality_analyzer_injection_helper",
"../../../api:peer_connection_quality_test_fixture_api",

123
test/pc/e2e/echo/echo_emulation.cc
View file

@ -1,123 +0,0 @@
/*
* Copyright (c) 2019 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/pc/e2e/echo/echo_emulation.h"
#include <limits>
#include <utility>
namespace webrtc {
namespace webrtc_pc_e2e {
namespace {
constexpr int kSingleBufferDurationMs = 10;
} // namespace
EchoEmulatingCapturer::EchoEmulatingCapturer(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
PeerConnectionE2EQualityTestFixture::EchoEmulationConfig config)
: delegate_(std::move(capturer)),
config_(config),
renderer_queue_(2 * config_.echo_delay.ms() / kSingleBufferDurationMs),
queue_input_(TestAudioDeviceModule::SamplesPerFrame(
delegate_->SamplingFrequency()) *
delegate_->NumChannels()),
queue_output_(TestAudioDeviceModule::SamplesPerFrame(
delegate_->SamplingFrequency()) *
delegate_->NumChannels()) {
renderer_thread_.Detach();
capturer_thread_.Detach();
}
void EchoEmulatingCapturer::OnAudioRendered(
rtc::ArrayView<const int16_t> data) {
RTC_DCHECK_RUN_ON(&renderer_thread_);
if (!recording_started_) {
// Because rendering can start before capturing in the beginning we can have
// a set of empty audio data frames. So we will skip them and will start
// fill the queue only after 1st non-empty audio data frame will arrive.
bool is_empty = true;
for (auto d : data) {
if (d != 0) {
is_empty = false;
break;
}
}
if (is_empty) {
return;
}
recording_started_ = true;
}
queue_input_.assign(data.begin(), data.end());
if (!renderer_queue_.Insert(&queue_input_)) {
// Test audio device works too slow with sanitizers and on some platforms
// and can't properly process audio, so when capturer will be stopped
// renderer will quickly overfill the queue.
// TODO(crbug.com/webrtc/10850) remove it when test ADM will be fast enough.
#if !defined(THREAD_SANITIZER) && !defined(MEMORY_SANITIZER) && \
!defined(ADDRESS_SANITIZER) && !defined(WEBRTC_ANDROID) && \
!(defined(_MSC_VER) && !defined(__clang__) && !defined(NDEBUG))
RTC_CHECK(false) << "Echo queue is full";
#endif
}
}
bool EchoEmulatingCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
RTC_DCHECK_RUN_ON(&capturer_thread_);
bool result = delegate_->Capture(buffer);
// Now we have to reduce input signal to avoid saturation when mixing in the
// fake echo.
for (size_t i = 0; i < buffer->size(); ++i) {
(*buffer)[i] /= 2;
}
// When we accumulated enough delay in the echo buffer we will pop from
// that buffer on each ::Capture(...) call. If the buffer become empty it
// will mean some bug, so we will crash during removing item from the queue.
if (!delay_accumulated_) {
delay_accumulated_ =
renderer_queue_.SizeAtLeast() >=
static_cast<size_t>(config_.echo_delay.ms() / kSingleBufferDurationMs);
}
if (delay_accumulated_) {
RTC_CHECK(renderer_queue_.Remove(&queue_output_));
for (size_t i = 0; i < buffer->size() && i < queue_output_.size(); ++i) {
int32_t res = (*buffer)[i] + queue_output_[i];
if (res < std::numeric_limits<int16_t>::min()) {
res = std::numeric_limits<int16_t>::min();
}
if (res > std::numeric_limits<int16_t>::max()) {
res = std::numeric_limits<int16_t>::max();
}
(*buffer)[i] = static_cast<int16_t>(res);
}
}
return result;
}
EchoEmulatingRenderer::EchoEmulatingRenderer(
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
EchoEmulatingCapturer* echo_emulating_capturer)
: delegate_(std::move(renderer)),
echo_emulating_capturer_(echo_emulating_capturer) {
RTC_DCHECK(echo_emulating_capturer_);
}
bool EchoEmulatingRenderer::Render(rtc::ArrayView<const int16_t> data) {
if (data.size() > 0) {
echo_emulating_capturer_->OnAudioRendered(data);
}
return delegate_->Render(data);
}
} // namespace webrtc_pc_e2e
} // namespace webrtc

79
test/pc/e2e/echo/echo_emulation.h
View file

@ -1,79 +0,0 @@
/*
* Copyright (c) 2019 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.
*/
#ifndef TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
#define TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
#include <atomic>
#include <deque>
#include <memory>
#include <vector>
#include "api/test/peerconnection_quality_test_fixture.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "rtc_base/swap_queue.h"
namespace webrtc {
namespace webrtc_pc_e2e {
// Reduces audio input strength from provided capturer twice and adds input
// provided into EchoEmulatingCapturer::OnAudioRendered(...).
class EchoEmulatingCapturer : public TestAudioDeviceModule::Capturer {
public:
EchoEmulatingCapturer(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
PeerConnectionE2EQualityTestFixture::EchoEmulationConfig config);
void OnAudioRendered(rtc::ArrayView<const int16_t> data);
int SamplingFrequency() const override {
return delegate_->SamplingFrequency();
}
int NumChannels() const override { return delegate_->NumChannels(); }
bool Capture(rtc::BufferT<int16_t>* buffer) override;
private:
std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_;
const PeerConnectionE2EQualityTestFixture::EchoEmulationConfig config_;
SwapQueue<std::vector<int16_t>> renderer_queue_;
SequenceChecker renderer_thread_;
std::vector<int16_t> queue_input_ RTC_GUARDED_BY(renderer_thread_);
bool recording_started_ RTC_GUARDED_BY(renderer_thread_) = false;
SequenceChecker capturer_thread_;
std::vector<int16_t> queue_output_ RTC_GUARDED_BY(capturer_thread_);
bool delay_accumulated_ RTC_GUARDED_BY(capturer_thread_) = false;
};
// Renders output into provided renderer and also copy output into provided
// EchoEmulationCapturer.
class EchoEmulatingRenderer : public TestAudioDeviceModule::Renderer {
public:
EchoEmulatingRenderer(
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
EchoEmulatingCapturer* echo_emulating_capturer);
int SamplingFrequency() const override {
return delegate_->SamplingFrequency();
}
int NumChannels() const override { return delegate_->NumChannels(); }
bool Render(rtc::ArrayView<const int16_t> data) override;
private:
std::unique_ptr<TestAudioDeviceModule::Renderer> delegate_;
EchoEmulatingCapturer* echo_emulating_capturer_;
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_ECHO_ECHO_EMULATION_H_

3
test/pc/e2e/peer_connection_e2e_smoke_test.cc
View file

@ -38,8 +38,6 @@ class PeerConnectionE2EQualityTestSmokeTest : public ::testing::Test {
using ScrollingParams = PeerConnectionE2EQualityTestFixture::ScrollingParams;
using VideoSimulcastConfig =
PeerConnectionE2EQualityTestFixture::VideoSimulcastConfig;
using EchoEmulationConfig =
PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
void RunTest(const std::string& test_case_name,
const RunParams& run_params,
@ -138,7 +136,6 @@ TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Smoke) {
run_params.use_flex_fec = true;
run_params.use_ulp_fec = true;
run_params.video_encoder_bitrate_multiplier = 1.1;
run_params.echo_emulation_config = EchoEmulationConfig();
RunTest(
"smoke", run_params,
[](PeerConfigurer* alice) {

4
test/pc/e2e/peer_connection_quality_test.cc
View file

@ -276,7 +276,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
[this]() { StartVideo(alice_video_sources_); }),
video_quality_analyzer_injection_helper_.get(), signaling_thread.get(),
alice_remote_audio_config, run_params.video_encoder_bitrate_multiplier,
run_params.echo_emulation_config, task_queue_.get());
task_queue_.get());
bob_ = TestPeer::CreateTestPeer(
std::move(bob_components), std::move(bob_params),
absl::make_unique<FixturePeerConnectionObserver>(
@ -287,7 +287,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
[this]() { StartVideo(bob_video_sources_); }),
video_quality_analyzer_injection_helper_.get(), signaling_thread.get(),
bob_remote_audio_config, run_params.video_encoder_bitrate_multiplier,
run_params.echo_emulation_config, task_queue_.get());
task_queue_.get());
int num_cores = CpuInfo::DetectNumberOfCores();
RTC_DCHECK_GE(num_cores, 1);

92
test/pc/e2e/test_peer.cc
View file

@ -26,7 +26,6 @@
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/client/basic_port_allocator.h"
#include "rtc_base/location.h"
#include "test/pc/e2e/echo/echo_emulation.h"
#include "test/testsupport/copy_to_file_audio_capturer.h"
namespace webrtc {
@ -37,8 +36,6 @@ using RemotePeerAudioConfig =
::webrtc::webrtc_pc_e2e::TestPeer::RemotePeerAudioConfig;
using AudioConfig =
::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::AudioConfig;
using EchoEmulationConfig = ::webrtc::webrtc_pc_e2e::
PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
constexpr int16_t kGeneratedAudioMaxAmplitude = 32000;
constexpr int kDefaultSamplingFrequencyInHz = 48000;
@ -75,15 +72,13 @@ class TestPeerComponents {
rtc::Thread* signaling_thread,
absl::optional<RemotePeerAudioConfig> remote_audio_config,
double bitrate_multiplier,
absl::optional<EchoEmulationConfig> echo_emulation_config,
rtc::TaskQueue* task_queue)
: audio_config_opt_(params.audio_config),
observer_(observer),
video_analyzer_helper_(video_analyzer_helper),
signaling_thread_(signaling_thread),
remote_audio_config_(std::move(remote_audio_config)),
bitrate_multiplier_(bitrate_multiplier),
echo_emulation_config_(std::move(echo_emulation_config)) {
bitrate_multiplier_(bitrate_multiplier) {
for (auto& video_config : params.video_configs) {
// Stream label should be set by fixture implementation here.
RTC_DCHECK(video_config.stream_label);
@ -182,26 +177,31 @@ class TestPeerComponents {
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
TaskQueueFactory* task_queue_factory) {
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
CreateAudioRenderer(remote_audio_config_);
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer =
CreateAudioCapturer(audio_config_opt_);
RTC_DCHECK(renderer);
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer;
if (audio_config_opt_) {
capturer = CreateAudioCapturer(*audio_config_opt_);
if (audio_config_opt_->input_dump_file_name) {
capturer = absl::make_unique<test::CopyToFileAudioCapturer>(
std::move(capturer),
audio_config_opt_->input_dump_file_name.value());
}
} else {
// If we have no audio config we still need to provide some audio device.
// In such case use generated capturer. Despite of we provided audio here,
// in test media setup audio stream won't be added into peer connection.
capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(
kGeneratedAudioMaxAmplitude, kDefaultSamplingFrequencyInHz);
}
RTC_DCHECK(capturer);
// Setup echo emulation if required.
if (echo_emulation_config_) {
capturer = absl::make_unique<EchoEmulatingCapturer>(
std::move(capturer), *echo_emulation_config_);
renderer = absl::make_unique<EchoEmulatingRenderer>(
std::move(renderer),
static_cast<EchoEmulatingCapturer*>(capturer.get()));
}
// Setup input stream dumping if required.
if (audio_config_opt_ && audio_config_opt_->input_dump_file_name) {
capturer = absl::make_unique<test::CopyToFileAudioCapturer>(
std::move(capturer), audio_config_opt_->input_dump_file_name.value());
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer;
if (remote_audio_config_ && remote_audio_config_->output_file_name) {
renderer = TestAudioDeviceModule::CreateBoundedWavFileWriter(
remote_audio_config_->output_file_name.value(),
remote_audio_config_->sampling_frequency_in_hz);
} else {
renderer = TestAudioDeviceModule::CreateDiscardRenderer(
kDefaultSamplingFrequencyInHz);
}
return TestAudioDeviceModule::Create(task_queue_factory,
@ -209,41 +209,19 @@ class TestPeerComponents {
std::move(renderer), /*speed=*/1.f);
}
std::unique_ptr<TestAudioDeviceModule::Renderer> CreateAudioRenderer(
const absl::optional<RemotePeerAudioConfig>& config) {
if (!config) {
// Return default renderer because we always require some renderer.
return TestAudioDeviceModule::CreateDiscardRenderer(
kDefaultSamplingFrequencyInHz);
}
if (config->output_file_name) {
return TestAudioDeviceModule::CreateBoundedWavFileWriter(
config->output_file_name.value(), config->sampling_frequency_in_hz);
}
return TestAudioDeviceModule::CreateDiscardRenderer(
config->sampling_frequency_in_hz);
}
std::unique_ptr<TestAudioDeviceModule::Capturer> CreateAudioCapturer(
const absl::optional<AudioConfig>& audio_config) {
if (!audio_config) {
// If we have no audio config we still need to provide some audio device.
// In such case use generated capturer. Despite of we provided audio here,
// in test media setup audio stream won't be added into peer connection.
const AudioConfig& audio_config) {
if (audio_config.mode == AudioConfig::Mode::kGenerated) {
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(
kGeneratedAudioMaxAmplitude, kDefaultSamplingFrequencyInHz);
kGeneratedAudioMaxAmplitude, audio_config.sampling_frequency_in_hz);
}
switch (audio_config->mode) {
case AudioConfig::Mode::kGenerated:
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(
kGeneratedAudioMaxAmplitude,
audio_config->sampling_frequency_in_hz);
case AudioConfig::Mode::kFile:
RTC_DCHECK(audio_config->input_file_name);
return TestAudioDeviceModule::CreateWavFileReader(
audio_config->input_file_name.value(), /*repeat=*/true);
if (audio_config.mode == AudioConfig::Mode::kFile) {
RTC_DCHECK(audio_config.input_file_name);
return TestAudioDeviceModule::CreateWavFileReader(
audio_config.input_file_name.value(), /*repeat=*/true);
}
RTC_NOTREACHED() << "Unknown audio_config->mode";
return nullptr;
}
std::unique_ptr<VideoEncoderFactory> CreateVideoEncoderFactory(
@ -312,7 +290,6 @@ class TestPeerComponents {
rtc::Thread* signaling_thread_;
absl::optional<RemotePeerAudioConfig> remote_audio_config_;
double bitrate_multiplier_;
absl::optional<EchoEmulationConfig> echo_emulation_config_;
};
} // namespace
@ -333,7 +310,6 @@ std::unique_ptr<TestPeer> TestPeer::CreateTestPeer(
rtc::Thread* signaling_thread,
absl::optional<RemotePeerAudioConfig> remote_audio_config,
double bitrate_multiplier,
absl::optional<EchoEmulationConfig> echo_emulation_config,
rtc::TaskQueue* task_queue) {
RTC_DCHECK(components);
RTC_DCHECK(params);
@ -343,7 +319,7 @@ std::unique_ptr<TestPeer> TestPeer::CreateTestPeer(
TestPeerComponents tpc(std::move(components), *params, observer.get(),
video_analyzer_helper, signaling_thread,
std::move(remote_audio_config), bitrate_multiplier,
echo_emulation_config, task_queue);
task_queue);
return absl::WrapUnique(new TestPeer(
tpc.peer_connection_factory(), tpc.peer_connection(), std::move(observer),

8
test/pc/e2e/test_peer.h
View file

@ -36,8 +36,6 @@ class TestPeer final : public PeerConnectionWrapper {
using PeerConnectionWrapper::PeerConnectionWrapper;
using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
using EchoEmulationConfig =
PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
struct RemotePeerAudioConfig {
RemotePeerAudioConfig(AudioConfig config)
@ -57,8 +55,11 @@ class TestPeer final : public PeerConnectionWrapper {
// injection.
//
// |signaling_thread| will be provided by test fixture implementation.
// |params| - describes current peer parameters, like current peer video
// |params| - describes current peer paramters, like current peer video
// streams and audio streams
// |audio_outpu_file_name| - the name of output file, where incoming audio
// stream should be written. It should be provided from remote peer
// |params.audio_config.output_file_name|
static std::unique_ptr<TestPeer> CreateTestPeer(
std::unique_ptr<InjectableComponents> components,
std::unique_ptr<Params> params,
@ -67,7 +68,6 @@ class TestPeer final : public PeerConnectionWrapper {
rtc::Thread* signaling_thread,
absl::optional<RemotePeerAudioConfig> remote_audio_config,
double bitrate_multiplier,
absl::optional<EchoEmulationConfig> echo_emulation_config,
rtc::TaskQueue* task_queue);
Params* params() const { return params_.get(); }