webrtc_m130/webrtc/modules/audio_processing/aec/system_delay_unittest.cc

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/*
* Copyright (c) 2012 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace {
class SystemDelayTest : public ::testing::Test {
protected:
SystemDelayTest();
virtual void SetUp();
virtual void TearDown();
// Initialization of AEC handle with respect to |sample_rate_hz|. Since the
// device sample rate is unimportant we set that value to 48000 Hz.
void Init(int sample_rate_hz);
// Makes one render call and one capture call in that specific order.
void RenderAndCapture(int device_buffer_ms);
// Fills up the far-end buffer with respect to the default device buffer size.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t BufferFillUp();
// Runs and verifies the behavior in a stable startup procedure.
void RunStableStartup();
// Maps buffer size in ms into samples, taking the unprocessed frame into
// account.
int MapBufferSizeToSamples(int size_in_ms, bool extended_filter);
void* handle_;
Aec* self_;
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t samples_per_frame_;
// Dummy input/output speech data.
static const int kSamplesPerChunk = 160;
float far_[kSamplesPerChunk];
float near_[kSamplesPerChunk];
float out_[kSamplesPerChunk];
const float* near_ptr_;
float* out_ptr_;
};
SystemDelayTest::SystemDelayTest()
: handle_(NULL), self_(NULL), samples_per_frame_(0) {
// Dummy input data are set with more or less arbitrary non-zero values.
for (int i = 0; i < kSamplesPerChunk; i++) {
far_[i] = 257.0;
near_[i] = 514.0;
}
memset(out_, 0, sizeof(out_));
near_ptr_ = near_;
out_ptr_ = out_;
}
void SystemDelayTest::SetUp() {
handle_ = WebRtcAec_Create();
ASSERT_TRUE(handle_);
self_ = reinterpret_cast<Aec*>(handle_);
}
void SystemDelayTest::TearDown() {
// Free AEC
WebRtcAec_Free(handle_);
handle_ = NULL;
}
// In SWB mode nothing is added to the buffer handling with respect to
// functionality compared to WB. We therefore only verify behavior in NB and WB.
static const int kSampleRateHz[] = {8000, 16000};
static const size_t kNumSampleRates =
sizeof(kSampleRateHz) / sizeof(*kSampleRateHz);
// Default audio device buffer size used.
static const int kDeviceBufMs = 100;
// Requirement for a stable device convergence time in ms. Should converge in
// less than |kStableConvergenceMs|.
static const int kStableConvergenceMs = 100;
// Maximum convergence time in ms. This means that we should leave the startup
// phase after |kMaxConvergenceMs| independent of device buffer stability
// conditions.
static const int kMaxConvergenceMs = 500;
void SystemDelayTest::Init(int sample_rate_hz) {
// Initialize AEC
EXPECT_EQ(0, WebRtcAec_Init(handle_, sample_rate_hz, 48000));
EXPECT_EQ(0, WebRtcAec_system_delay(self_->aec));
// One frame equals 10 ms of data.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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samples_per_frame_ = static_cast<size_t>(sample_rate_hz / 100);
}
void SystemDelayTest::RenderAndCapture(int device_buffer_ms) {
EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
EXPECT_EQ(0,
WebRtcAec_Process(handle_,
&near_ptr_,
1,
&out_ptr_,
samples_per_frame_,
device_buffer_ms,
0));
}
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t SystemDelayTest::BufferFillUp() {
// To make sure we have a full buffer when we verify stability we first fill
// up the far-end buffer with the same amount as we will report in through
// Process().
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t buffer_size = 0;
for (int i = 0; i < kDeviceBufMs / 10; i++) {
EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
buffer_size += samples_per_frame_;
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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EXPECT_EQ(static_cast<int>(buffer_size),
WebRtcAec_system_delay(self_->aec));
}
return buffer_size;
}
void SystemDelayTest::RunStableStartup() {
// To make sure we have a full buffer when we verify stability we first fill
// up the far-end buffer with the same amount as we will report in through
// Process().
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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size_t buffer_size = BufferFillUp();
if (WebRtcAec_delay_agnostic_enabled(self_->aec) == 1) {
// In extended_filter mode we set the buffer size after the first processed
// 10 ms chunk. Hence, we don't need to wait for the reported system delay
// values to become stable.
RenderAndCapture(kDeviceBufMs);
buffer_size += samples_per_frame_;
EXPECT_EQ(0, self_->startup_phase);
} else {
// A stable device should be accepted and put in a regular process mode
// within |kStableConvergenceMs|.
int process_time_ms = 0;
for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
RenderAndCapture(kDeviceBufMs);
buffer_size += samples_per_frame_;
if (self_->startup_phase == 0) {
// We have left the startup phase.
break;
}
}
// Verify convergence time.
EXPECT_GT(kStableConvergenceMs, process_time_ms);
}
// Verify that the buffer has been flushed.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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EXPECT_GE(static_cast<int>(buffer_size),
WebRtcAec_system_delay(self_->aec));
}
int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms,
bool extended_filter) {
// If extended_filter is disabled we add an extra 10 ms for the unprocessed
// frame. That is simply how the algorithm is constructed.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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return static_cast<int>(
(size_in_ms + (extended_filter ? 0 : 10)) * samples_per_frame_ / 10);
}
// The tests should meet basic requirements and not be adjusted to what is
// actually implemented. If we don't get good code coverage this way we either
// lack in tests or have unnecessary code.
// General requirements:
// 1) If we add far-end data the system delay should be increased with the same
// amount we add.
// 2) If the far-end buffer is full we should flush the oldest data to make room
// for the new. In this case the system delay is unaffected.
// 3) There should exist a startup phase in which the buffer size is to be
// determined. In this phase no cancellation should be performed.
// 4) Under stable conditions (small variations in device buffer sizes) the AEC
// should determine an appropriate local buffer size within
// |kStableConvergenceMs| ms.
// 5) Under unstable conditions the AEC should make a decision within
// |kMaxConvergenceMs| ms.
// 6) If the local buffer runs out of data we should stuff the buffer with older
// frames.
// 7) The system delay should within |kMaxConvergenceMs| ms heal from
// disturbances like drift, data glitches, toggling events and outliers.
// 8) The system delay should never become negative.
TEST_F(SystemDelayTest, CorrectIncreaseWhenBufferFarend) {
// When we add data to the AEC buffer the internal system delay should be
// incremented with the same amount as the size of data.
// This process should be independent of DA-AEC and extended_filter mode.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
// Loop through a couple of calls to make sure the system delay
// increments correctly.
for (int j = 1; j <= 5; j++) {
EXPECT_EQ(0,
WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
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EXPECT_EQ(static_cast<int>(j * samples_per_frame_),
WebRtcAec_system_delay(self_->aec));
}
}
}
}
}
// TODO(bjornv): Add a test to verify behavior if the far-end buffer is full
// when adding new data.
TEST_F(SystemDelayTest, CorrectDelayAfterStableStartup) {
// We run the system in a stable startup. After that we verify that the system
// delay meets the requirements.
// This process should be independent of DA-AEC and extended_filter mode.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
// Verify system delay with respect to requirements, i.e., the
// |system_delay| is in the interval [75%, 100%] of what's reported on
// the average.
// In extended_filter mode we target 50% and measure after one processed
// 10 ms chunk.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
int average_reported_delay =
static_cast<int>(kDeviceBufMs * samples_per_frame_ / 10);
EXPECT_GE(average_reported_delay, WebRtcAec_system_delay(self_->aec));
int lower_bound = WebRtcAec_extended_filter_enabled(self_->aec)
? average_reported_delay / 2 - samples_per_frame_
: average_reported_delay * 3 / 4;
EXPECT_LE(lower_bound, WebRtcAec_system_delay(self_->aec));
}
}
}
}
TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) {
// This test does not apply in extended_filter mode, since we only use the
// the first 10 ms chunk to determine a reasonable buffer size. Neither does
// it apply if DA-AEC is on because that overrides the startup procedure.
WebRtcAec_enable_extended_filter(self_->aec, 0);
EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(self_->aec));
WebRtcAec_enable_delay_agnostic(self_->aec, 0);
EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(self_->aec));
// In an unstable system we would start processing after |kMaxConvergenceMs|.
// On the last frame the AEC buffer is adjusted to 60% of the last reported
// device buffer size.
// We construct an unstable system by altering the device buffer size between
// two values |kDeviceBufMs| +- 25 ms.
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
// To make sure we have a full buffer when we verify stability we first fill
// up the far-end buffer with the same amount as we will report in on the
// average through Process().
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t buffer_size = BufferFillUp();
int buffer_offset_ms = 25;
int reported_delay_ms = 0;
int process_time_ms = 0;
for (; process_time_ms <= kMaxConvergenceMs; process_time_ms += 10) {
reported_delay_ms = kDeviceBufMs + buffer_offset_ms;
RenderAndCapture(reported_delay_ms);
buffer_size += samples_per_frame_;
buffer_offset_ms = -buffer_offset_ms;
Add an extended filter mode to AEC. Re-land: http://review.webrtc.org/2151007/ TBR=bjornv@webrtc.org Original change description: This mode extends the filter length from the current 48 ms to 128 ms. It is runtime selectable which allows it to be enabled through experiment. We reuse the DelayCorrection infrastructure to avoid having to replumb everything up to libjingle. Increases AEC complexity by ~50% on modern x86 CPUs. Measurements (in percent of usage on one core): Machine/CPU Normal Extended MacBook Retina (Early 2013), Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9% MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7% Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0% Samsung ARM Chromebook, Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6% The relative value is large of course but the absolute should be acceptable in order to have a working AEC on some platforms. Detailed changes to the algorithm: - The filter length is changed from 48 to 128 ms. This comes with tuning of several parameters: i) filter adaptation stepsize and error threshold; ii) non-linear processing smoothing and overdrive. - Option to ignore the reported delays on platforms which we deem sufficiently unreliable. Currently this will be enabled in Chromium for Mac. - Faster startup times by removing the excessive "startup phase" processing of reported delays. - Much more conservative adjustments to the far-end read pointer. We smooth the delay difference more heavily, and back off from the difference more. Adjustments force a readaptation of the filter, so they should be avoided except when really necessary. Corresponds to these changes: https://chromereviews.googleplex.com/9412014 https://chromereviews.googleplex.com/9514013 https://chromereviews.googleplex.com/9960013 BUG=454,827,1261 Review URL: https://webrtc-codereview.appspot.com/2295006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:17:38 +00:00
if (self_->startup_phase == 0) {
// We have left the startup phase.
break;
}
}
// Verify convergence time.
EXPECT_GE(kMaxConvergenceMs, process_time_ms);
// Verify that the buffer has been flushed.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
EXPECT_GE(static_cast<int>(buffer_size),
WebRtcAec_system_delay(self_->aec));
// Verify system delay with respect to requirements, i.e., the
// |system_delay| is in the interval [60%, 100%] of what's last reported.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
EXPECT_GE(static_cast<int>(reported_delay_ms * samples_per_frame_ / 10),
WebRtcAec_system_delay(self_->aec));
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
EXPECT_LE(
static_cast<int>(reported_delay_ms * samples_per_frame_ / 10 * 3 / 5),
WebRtcAec_system_delay(self_->aec));
}
}
TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) {
// This test does not apply in extended_filter mode, since we only use the
// the first 10 ms chunk to determine a reasonable buffer size. Neither does
// it apply if DA-AEC is on because that overrides the startup procedure.
WebRtcAec_enable_extended_filter(self_->aec, 0);
EXPECT_EQ(0, WebRtcAec_extended_filter_enabled(self_->aec));
WebRtcAec_enable_delay_agnostic(self_->aec, 0);
EXPECT_EQ(0, WebRtcAec_delay_agnostic_enabled(self_->aec));
// In this test we start by establishing the device buffer size during stable
// conditions, but with an empty internal far-end buffer. Once that is done we
// verify that the system delay is increased correctly until we have reach an
// internal buffer size of 75% of what's been reported.
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
// We assume that running |kStableConvergenceMs| calls will put the
// algorithm in a state where the device buffer size has been determined. We
// can make that assumption since we have a separate stability test.
int process_time_ms = 0;
for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
EXPECT_EQ(0,
WebRtcAec_Process(handle_,
&near_ptr_,
1,
&out_ptr_,
samples_per_frame_,
kDeviceBufMs,
0));
}
// Verify that a buffer size has been established.
EXPECT_EQ(0, self_->checkBuffSize);
// We now have established the required buffer size. Let us verify that we
// fill up before leaving the startup phase for normal processing.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t buffer_size = 0;
size_t target_buffer_size = kDeviceBufMs * samples_per_frame_ / 10 * 3 / 4;
process_time_ms = 0;
for (; process_time_ms <= kMaxConvergenceMs; process_time_ms += 10) {
RenderAndCapture(kDeviceBufMs);
buffer_size += samples_per_frame_;
if (self_->startup_phase == 0) {
// We have left the startup phase.
break;
}
}
// Verify convergence time.
EXPECT_GT(kMaxConvergenceMs, process_time_ms);
// Verify that the buffer has reached the desired size.
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
EXPECT_LE(static_cast<int>(target_buffer_size),
WebRtcAec_system_delay(self_->aec));
// Verify normal behavior (system delay is kept constant) after startup by
// running a couple of calls to BufferFarend() and Process().
for (int j = 0; j < 6; j++) {
int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
RenderAndCapture(kDeviceBufMs);
EXPECT_EQ(system_delay_before_calls, WebRtcAec_system_delay(self_->aec));
}
}
}
TEST_F(SystemDelayTest, CorrectDelayWhenBufferUnderrun) {
// Here we test a buffer under run scenario. If we keep on calling
// WebRtcAec_Process() we will finally run out of data, but should
// automatically stuff the buffer. We verify this behavior by checking if the
// system delay goes negative.
// This process should be independent of DA-AEC and extended_filter mode.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
// The AEC has now left the Startup phase. We now have at most
// |kStableConvergenceMs| in the buffer. Keep on calling Process() until
// we run out of data and verify that the system delay is non-negative.
for (int j = 0; j <= kStableConvergenceMs; j += 10) {
EXPECT_EQ(0, WebRtcAec_Process(handle_, &near_ptr_, 1, &out_ptr_,
samples_per_frame_, kDeviceBufMs, 0));
EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
}
}
}
}
}
TEST_F(SystemDelayTest, CorrectDelayDuringDrift) {
// This drift test should verify that the system delay is never exceeding the
// device buffer. The drift is simulated by decreasing the reported device
// buffer size by 1 ms every 100 ms. If the device buffer size goes below 30
// ms we jump (add) 10 ms to give a repeated pattern.
// This process should be independent of DA-AEC and extended_filter mode.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
// We have left the startup phase and proceed with normal processing.
int jump = 0;
for (int j = 0; j < 1000; j++) {
// Drift = -1 ms per 100 ms of data.
int device_buf_ms = kDeviceBufMs - (j / 10) + jump;
int device_buf = MapBufferSizeToSamples(device_buf_ms,
extended_filter == 1);
if (device_buf_ms < 30) {
// Add 10 ms data, taking affect next frame.
jump += 10;
}
RenderAndCapture(device_buf_ms);
// Verify that the system delay does not exceed the device buffer.
EXPECT_GE(device_buf, WebRtcAec_system_delay(self_->aec));
// Verify that the system delay is non-negative.
EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
}
}
}
}
}
TEST_F(SystemDelayTest, ShouldRecoverAfterGlitch) {
// This glitch test should verify that the system delay recovers if there is
// a glitch in data. The data glitch is constructed as 200 ms of buffering
// after which the stable procedure continues. The glitch is never reported by
// the device.
// The system is said to be in a non-causal state if the difference between
// the device buffer and system delay is less than a block (64 samples).
// This process should be independent of DA-AEC and extended_filter mode.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
extended_filter == 1);
// Glitch state.
for (int j = 0; j < 20; j++) {
EXPECT_EQ(0,
WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
// No need to verify system delay, since that is done in a separate
// test.
}
// Verify that we are in a non-causal state, i.e.,
// |system_delay| > |device_buf|.
EXPECT_LT(device_buf, WebRtcAec_system_delay(self_->aec));
// Recover state. Should recover at least 4 ms of data per 10 ms, hence
// a glitch of 200 ms will take at most 200 * 10 / 4 = 500 ms to recover
// from.
bool non_causal = true; // We are currently in a non-causal state.
for (int j = 0; j < 50; j++) {
int system_delay_before = WebRtcAec_system_delay(self_->aec);
RenderAndCapture(kDeviceBufMs);
int system_delay_after = WebRtcAec_system_delay(self_->aec);
// We have recovered if
// |device_buf| - |system_delay_after| >= PART_LEN (1 block).
// During recovery, |system_delay_after| < |system_delay_before|,
// otherwise they are equal.
if (non_causal) {
EXPECT_LT(system_delay_after, system_delay_before);
if (device_buf - system_delay_after >= PART_LEN) {
non_causal = false;
}
} else {
EXPECT_EQ(system_delay_before, system_delay_after);
}
// Verify that the system delay is non-negative.
EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
}
// Check that we have recovered.
EXPECT_FALSE(non_causal);
}
}
}
}
TEST_F(SystemDelayTest, UnaffectedWhenSpuriousDeviceBufferValues) {
// This test does not apply in extended_filter mode, since we only use the
// the first 10 ms chunk to determine a reasonable buffer size.
const int extended_filter = 0;
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
// Should be DA-AEC independent.
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
// This spurious device buffer data test aims at verifying that the system
// delay is unaffected by large outliers.
// The system is said to be in a non-causal state if the difference between
// the device buffer and system delay is less than a block (64 samples).
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
extended_filter == 1);
// Normal state. We are currently not in a non-causal state.
bool non_causal = false;
// Run 1 s and replace device buffer size with 500 ms every 100 ms.
for (int j = 0; j < 100; j++) {
int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
int device_buf_ms = j % 10 == 0 ? 500 : kDeviceBufMs;
RenderAndCapture(device_buf_ms);
// Check for non-causality.
if (device_buf - WebRtcAec_system_delay(self_->aec) < PART_LEN) {
non_causal = true;
}
EXPECT_FALSE(non_causal);
EXPECT_EQ(system_delay_before_calls,
WebRtcAec_system_delay(self_->aec));
// Verify that the system delay is non-negative.
EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
}
}
}
}
TEST_F(SystemDelayTest, CorrectImpactWhenTogglingDeviceBufferValues) {
// This test aims at verifying that the system delay is "unaffected" by
// toggling values reported by the device.
// The test is constructed such that every other device buffer value is zero
// and then 2 * |kDeviceBufMs|, hence the size is constant on the average. The
// zero values will force us into a non-causal state and thereby lowering the
// system delay until we basically run out of data. Once that happens the
// buffer will be stuffed.
// TODO(bjornv): This test will have a better impact if we verified that the
// delay estimate goes up when the system delay goes down to meet the average
// device buffer size.
// This test does not apply if DA-AEC is enabled and extended_filter mode
// disabled.
for (int extended_filter = 0; extended_filter <= 1; ++extended_filter) {
WebRtcAec_enable_extended_filter(self_->aec, extended_filter);
EXPECT_EQ(extended_filter, WebRtcAec_extended_filter_enabled(self_->aec));
for (int da_aec = 0; da_aec <= 1; ++da_aec) {
WebRtcAec_enable_delay_agnostic(self_->aec, da_aec);
EXPECT_EQ(da_aec, WebRtcAec_delay_agnostic_enabled(self_->aec));
if (extended_filter == 0 && da_aec == 1) {
continue;
}
for (size_t i = 0; i < kNumSampleRates; i++) {
Init(kSampleRateHz[i]);
RunStableStartup();
const int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
extended_filter == 1);
// Normal state. We are currently not in a non-causal state.
bool non_causal = false;
// Loop through 100 frames (both render and capture), which equals 1 s
// of data. Every odd frame we set the device buffer size to
// 2 * |kDeviceBufMs| and even frames we set the device buffer size to
// zero.
for (int j = 0; j < 100; j++) {
int system_delay_before_calls = WebRtcAec_system_delay(self_->aec);
int device_buf_ms = 2 * (j % 2) * kDeviceBufMs;
RenderAndCapture(device_buf_ms);
// Check for non-causality, compared with the average device buffer
// size.
non_causal |= (device_buf - WebRtcAec_system_delay(self_->aec) < 64);
EXPECT_GE(system_delay_before_calls,
WebRtcAec_system_delay(self_->aec));
// Verify that the system delay is non-negative.
EXPECT_LE(0, WebRtcAec_system_delay(self_->aec));
}
// Verify we are not in a non-causal state.
EXPECT_FALSE(non_causal);
}
}
}
}
} // namespace
} // namespace webrtc