From 7056908774c2f00e6c0683898b86811fa828bfc5 Mon Sep 17 00:00:00 2001 From: "bjornv@webrtc.org" Date: Thu, 12 Apr 2012 12:13:50 +0000 Subject: [PATCH] System delay unit tests Added a system delay test class. Noticed I don't need the ApmTest class at all, which simplified the implementation. Start at patch set 3. The others are not complete. BUG=None TEST= Review URL: https://webrtc-codereview.appspot.com/475003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2014 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/modules/audio_processing/aec/aec.gypi | 1 + .../audio_processing/aec/echo_cancellation.c | 56 +-- .../aec/echo_cancellation_internal.h | 67 +++ .../audio_processing/aec/system_delay_test.cc | 459 ++++++++++++++++++ src/modules/audio_processing/apm_tests.gypi | 7 +- 5 files changed, 536 insertions(+), 54 deletions(-) create mode 100644 src/modules/audio_processing/aec/echo_cancellation_internal.h create mode 100644 src/modules/audio_processing/aec/system_delay_test.cc diff --git a/src/modules/audio_processing/aec/aec.gypi b/src/modules/audio_processing/aec/aec.gypi index 2e34c5bb8e..1506342856 100644 --- a/src/modules/audio_processing/aec/aec.gypi +++ b/src/modules/audio_processing/aec/aec.gypi @@ -30,6 +30,7 @@ 'sources': [ 'include/echo_cancellation.h', 'echo_cancellation.c', + 'echo_cancellation_internal.h', 'aec_core.h', 'aec_core.c', 'aec_rdft.h', diff --git a/src/modules/audio_processing/aec/echo_cancellation.c b/src/modules/audio_processing/aec/echo_cancellation.c index 5d690b9e62..69ff568932 100644 --- a/src/modules/audio_processing/aec/echo_cancellation.c +++ b/src/modules/audio_processing/aec/echo_cancellation.c @@ -23,6 +23,7 @@ #include "aec_core.h" #include "aec_resampler.h" #include "common_audio/signal_processing/include/signal_processing_library.h" +#include "modules/audio_processing/aec/echo_cancellation_internal.h" #include "ring_buffer.h" #include "typedefs.h" @@ -44,58 +45,6 @@ static const int initCheck = 42; static int instance_count = 0; #endif -typedef struct { - int delayCtr; - int sampFreq; - int splitSampFreq; - int scSampFreq; - float sampFactor; // scSampRate / sampFreq - short nlpMode; - short autoOnOff; - short activity; - short skewMode; - int bufSizeStart; - //short bufResetCtr; // counts number of noncausal frames - int knownDelay; - - short initFlag; // indicates if AEC has been initialized - - // Variables used for averaging far end buffer size - short counter; - int sum; - short firstVal; - short checkBufSizeCtr; - - // Variables used for delay shifts - short msInSndCardBuf; - short filtDelay; // Filtered delay estimate. - int timeForDelayChange; - int ECstartup; - int checkBuffSize; - short lastDelayDiff; - -#ifdef WEBRTC_AEC_DEBUG_DUMP - void* far_pre_buf_s16; // Time domain far-end pre-buffer in int16_t. - FILE *bufFile; - FILE *delayFile; - FILE *skewFile; -#endif - - // Structures - void *resampler; - - int skewFrCtr; - int resample; // if the skew is small enough we don't resample - int highSkewCtr; - float skew; - - void* far_pre_buf; // Time domain far-end pre-buffer. - - int lastError; - - aec_t *aec; -} aecpc_t; - // Estimates delay to set the position of the far-end buffer read pointer // (controlled by knownDelay) static int EstBufDelay(aecpc_t *aecInst); @@ -880,6 +829,8 @@ static int EstBufDelay(aecpc_t* aecpc) { // Before we proceed with the delay estimate filtering we: // 1) Compensate for the frame that will be read. // 2) Compensate for drift resampling. + // 3) Compensate for non-causality if needed, since the estimated delay can't + // be negative. // 1) Compensating for the frame(s) that will be read/processed. current_delay += FRAME_LEN * aecpc->aec->mult; @@ -889,6 +840,7 @@ static int EstBufDelay(aecpc_t* aecpc) { current_delay -= kResamplingDelay; } + // 3) Compensate for non-causality, if needed, by flushing one block. if (current_delay < PART_LEN) { current_delay += WebRtcAec_MoveFarReadPtr(aecpc->aec, 1) * PART_LEN; } diff --git a/src/modules/audio_processing/aec/echo_cancellation_internal.h b/src/modules/audio_processing/aec/echo_cancellation_internal.h new file mode 100644 index 0000000000..b218fce9b8 --- /dev/null +++ b/src/modules/audio_processing/aec/echo_cancellation_internal.h @@ -0,0 +1,67 @@ +/* + * 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. + */ + +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_ + +#include "modules/audio_processing/aec/aec_core.h" + +typedef struct { + int delayCtr; + int sampFreq; + int splitSampFreq; + int scSampFreq; + float sampFactor; // scSampRate / sampFreq + short nlpMode; + short autoOnOff; + short activity; + short skewMode; + int bufSizeStart; + int knownDelay; + + short initFlag; // indicates if AEC has been initialized + + // Variables used for averaging far end buffer size + short counter; + int sum; + short firstVal; + short checkBufSizeCtr; + + // Variables used for delay shifts + short msInSndCardBuf; + short filtDelay; // Filtered delay estimate. + int timeForDelayChange; + int ECstartup; + int checkBuffSize; + short lastDelayDiff; + +#ifdef WEBRTC_AEC_DEBUG_DUMP + void* far_pre_buf_s16; // Time domain far-end pre-buffer in int16_t. + FILE* bufFile; + FILE* delayFile; + FILE* skewFile; +#endif + + // Structures + void* resampler; + + int skewFrCtr; + int resample; // if the skew is small enough we don't resample + int highSkewCtr; + float skew; + + void* far_pre_buf; // Time domain far-end pre-buffer. + + int lastError; + + aec_t* aec; +} aecpc_t; + +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_ diff --git a/src/modules/audio_processing/aec/system_delay_test.cc b/src/modules/audio_processing/aec/system_delay_test.cc new file mode 100644 index 0000000000..272cb8a78c --- /dev/null +++ b/src/modules/audio_processing/aec/system_delay_test.cc @@ -0,0 +1,459 @@ +/* + * 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 "gtest/gtest.h" + +#include "modules/audio_processing/aec/include/echo_cancellation.h" +#include "modules/audio_processing/aec/echo_cancellation_internal.h" +#include "typedefs.h" + +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. + int 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); + + void* handle_; + aecpc_t* self_; + int samples_per_frame_; + // Dummy input/output speech data. + int16_t far_[160]; + int16_t near_[160]; + int16_t out_[160]; +}; + +SystemDelayTest::SystemDelayTest() + : handle_(NULL), + self_(NULL), + samples_per_frame_(0) { + // Dummy input data are set with more or less arbitrary non-zero values. + memset(far_, 1, sizeof(far_)); + memset(near_, 2, sizeof(near_)); + memset(out_, 0, sizeof(out_)); +} + +void SystemDelayTest::SetUp() { + ASSERT_EQ(0, WebRtcAec_Create(&handle_)); + self_ = reinterpret_cast(handle_); +} + +void SystemDelayTest::TearDown() { + // Free AEC + ASSERT_EQ(0, 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)); + + // One frame equals 10 ms of data. + samples_per_frame_ = 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_, NULL, out_, NULL, + samples_per_frame_, device_buffer_ms, 0)); +} + +int 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(). + int 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_; + EXPECT_EQ(buffer_size, self_->aec->system_delay); + } + 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(). + int buffer_size = BufferFillUp(); + // 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_->ECstartup == 0) { + // We have left the startup phase. + break; + } + } + // Verify convergence time. + EXPECT_GT(kStableConvergenceMs, process_time_ms); + // Verify that the buffer has been flushed. + EXPECT_GE(buffer_size, self_->aec->system_delay); +} + +int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms) { + // The extra 10 ms corresponds to the unprocessed frame. + return (size_in_ms + 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. + 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_)); + EXPECT_EQ(j * samples_per_frame_, self_->aec->system_delay); + } + } +} + +// 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. + 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. + int average_reported_delay = kDeviceBufMs * samples_per_frame_ / 10; + EXPECT_GE(average_reported_delay, self_->aec->system_delay); + EXPECT_LE(average_reported_delay * 3 / 4, self_->aec->system_delay); + } +} + +TEST_F(SystemDelayTest, CorrectDelayAfterUnstableStartup) { + // 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(). + int 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; + if (self_->ECstartup == 0) { + // We have left the startup phase. + break; + } + } + // Verify convergence time. + EXPECT_GE(kMaxConvergenceMs, process_time_ms); + // Verify that the buffer has been flushed. + EXPECT_GE(buffer_size, self_->aec->system_delay); + + // Verify system delay with respect to requirements, i.e., the + // |system_delay| is in the interval [60%, 100%] of what's last reported. + EXPECT_GE(reported_delay_ms * samples_per_frame_ / 10, + self_->aec->system_delay); + EXPECT_LE(reported_delay_ms * samples_per_frame_ / 10 * 3 / 5, + self_->aec->system_delay); + } +} + +TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) { + // 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_, NULL, out_, NULL, + 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. + int buffer_size = 0; + int 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_->ECstartup == 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. + EXPECT_LE(target_buffer_size, self_->aec->system_delay); + + // 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 = self_->aec->system_delay; + RenderAndCapture(kDeviceBufMs); + EXPECT_EQ(system_delay_before_calls, self_->aec->system_delay); + } + } +} + +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. + 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_, NULL, out_, NULL, + samples_per_frame_, kDeviceBufMs, 0)); + EXPECT_LE(0, self_->aec->system_delay); + } + } +} + +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. + for (size_t i = 0; i < kNumSampleRates; i++) { + Init(kSampleRateHz[i]); + RunStableStartup(); + + // We have now 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); + + 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, self_->aec->system_delay); + + // Verify that the system delay is non-negative. + EXPECT_LE(0, self_->aec->system_delay); + } + } +} + +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). + for (size_t i = 0; i < kNumSampleRates; i++) { + Init(kSampleRateHz[i]); + RunStableStartup(); + int device_buf = MapBufferSizeToSamples(kDeviceBufMs); + // 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, self_->aec->system_delay); + + // 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 = self_->aec->system_delay; + RenderAndCapture(kDeviceBufMs); + int system_delay_after = self_->aec->system_delay; + + // We have recovered if |device_buf| - |system_delay_after| >= 64 (one + // 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 >= 64) { + non_causal = false; + } + } else { + EXPECT_EQ(system_delay_before, system_delay_after); + } + // Verify that the system delay is non-negative. + EXPECT_LE(0, self_->aec->system_delay); + } + // Check that we have recovered. + EXPECT_FALSE(non_causal); + } +} + +TEST_F(SystemDelayTest, UnaffectedWhenSpuriousDeviceBufferValues) { + // 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); + + // 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 = self_->aec->system_delay; + int device_buf_ms = kDeviceBufMs; + if (j % 10 == 0) { + device_buf_ms = 500; + } + RenderAndCapture(device_buf_ms); + + // Check for non-causality. + if (device_buf - self_->aec->system_delay < 64) { + non_causal = true; + } + EXPECT_FALSE(non_causal); + EXPECT_EQ(system_delay_before_calls, self_->aec->system_delay); + + // Verify that the system delay is non-negative. + EXPECT_LE(0, self_->aec->system_delay); + } + } +} + +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 runs 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 done to meet the average + // device buffer size. + for (size_t i = 0; i < kNumSampleRates; i++) { + Init(kSampleRateHz[i]); + RunStableStartup(); + int device_buf = MapBufferSizeToSamples(kDeviceBufMs); + + // 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 = self_->aec->system_delay; + 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 - self_->aec->system_delay < 64); + EXPECT_GE(system_delay_before_calls, self_->aec->system_delay); + + // Verify that the system delay is non-negative. + EXPECT_LE(0, self_->aec->system_delay); + } + // Verify we are not in a non-causal state. + EXPECT_FALSE(non_causal); + } +} + +} // namespace diff --git a/src/modules/audio_processing/apm_tests.gypi b/src/modules/audio_processing/apm_tests.gypi index c00a6d2732..2b85d0ff98 100644 --- a/src/modules/audio_processing/apm_tests.gypi +++ b/src/modules/audio_processing/apm_tests.gypi @@ -1,4 +1,4 @@ -# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. +# 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 @@ -29,7 +29,10 @@ '<(webrtc_root)/../test/test.gyp:test_support', '<(webrtc_root)/../testing/gtest.gyp:gtest', ], - 'sources': [ 'test/unit_test.cc', ], + 'sources': [ + 'aec/system_delay_test.cc', + 'test/unit_test.cc', + ], }, { 'target_name': 'audioproc_unittest_proto',