Revert "Introduce ability to test echo in PC level test framework"
This reverts commit 77acb015b6ba886da3e7adb9c2106cf873fa8497.
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}
diff --git a/api/test/peerconnection_quality_test_fixture.h b/api/test/peerconnection_quality_test_fixture.h
index 44bb1f0..12907bc 100644
--- a/api/test/peerconnection_quality_test_fixture.h
+++ b/api/test/peerconnection_quality_test_fixture.h
@@ -279,13 +279,6 @@
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 @@
// 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.
diff --git a/rtc_base/swap_queue.h b/rtc_base/swap_queue.h
index eb0b1ff..8914548 100644
--- a/rtc_base/swap_queue.h
+++ b/rtc_base/swap_queue.h
@@ -200,16 +200,6 @@
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.
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 440064b..6d24bbb 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -203,20 +203,6 @@
]
}
- 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 @@
"test_peer.h",
]
deps = [
- ":echo_emulation",
":peer_connection_quality_test_params",
":video_quality_analyzer_injection_helper",
"../../../api:peer_connection_quality_test_fixture_api",
diff --git a/test/pc/e2e/echo/echo_emulation.cc b/test/pc/e2e/echo/echo_emulation.cc
deleted file mode 100644
index 1405570..0000000
--- a/test/pc/e2e/echo/echo_emulation.cc
+++ /dev/null
@@ -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
diff --git a/test/pc/e2e/echo/echo_emulation.h b/test/pc/e2e/echo/echo_emulation.h
deleted file mode 100644
index d1d41f6..0000000
--- a/test/pc/e2e/echo/echo_emulation.h
+++ /dev/null
@@ -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_
diff --git a/test/pc/e2e/peer_connection_e2e_smoke_test.cc b/test/pc/e2e/peer_connection_e2e_smoke_test.cc
index dc1d819..a6f4b5e 100644
--- a/test/pc/e2e/peer_connection_e2e_smoke_test.cc
+++ b/test/pc/e2e/peer_connection_e2e_smoke_test.cc
@@ -38,8 +38,6 @@
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 @@
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) {
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc
index c623cc9..e90b170 100644
--- a/test/pc/e2e/peer_connection_quality_test.cc
+++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -276,7 +276,7 @@
[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 @@
[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);
diff --git a/test/pc/e2e/test_peer.cc b/test/pc/e2e/test_peer.cc
index 6cc1168..0e044b4 100644
--- a/test/pc/e2e/test_peer.cc
+++ b/test/pc/e2e/test_peer.cc
@@ -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 @@
::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 @@
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 @@
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 @@
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 @@
rtc::Thread* signaling_thread_;
absl::optional<RemotePeerAudioConfig> remote_audio_config_;
double bitrate_multiplier_;
- absl::optional<EchoEmulationConfig> echo_emulation_config_;
};
} // namespace
@@ -333,7 +310,6 @@
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 @@
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),
diff --git a/test/pc/e2e/test_peer.h b/test/pc/e2e/test_peer.h
index efacde5..8cb8415 100644
--- a/test/pc/e2e/test_peer.h
+++ b/test/pc/e2e/test_peer.h
@@ -36,8 +36,6 @@
using PeerConnectionWrapper::PeerConnectionWrapper;
using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
- using EchoEmulationConfig =
- PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
struct RemotePeerAudioConfig {
RemotePeerAudioConfig(AudioConfig config)
@@ -57,8 +55,11 @@
// 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 @@
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(); }
