mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
d47941e018
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
439 lines
18 KiB
C++
439 lines
18 KiB
C++
/*
|
|
* Copyright (c) 2016 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 <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/audio_buffer.h"
|
|
#include "modules/audio_processing/gain_control_impl.h"
|
|
#include "modules/audio_processing/test/audio_buffer_tools.h"
|
|
#include "modules/audio_processing/test/bitexactness_tools.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
const int kNumFramesToProcess = 100;
|
|
|
|
void ProcessOneFrame(int sample_rate_hz,
|
|
AudioBuffer* render_audio_buffer,
|
|
AudioBuffer* capture_audio_buffer,
|
|
GainControlImpl* gain_controller) {
|
|
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
|
|
render_audio_buffer->SplitIntoFrequencyBands();
|
|
capture_audio_buffer->SplitIntoFrequencyBands();
|
|
}
|
|
|
|
std::vector<int16_t> render_audio;
|
|
GainControlImpl::PackRenderAudioBuffer(render_audio_buffer, &render_audio);
|
|
gain_controller->ProcessRenderAudio(render_audio);
|
|
gain_controller->AnalyzeCaptureAudio(capture_audio_buffer);
|
|
gain_controller->ProcessCaptureAudio(capture_audio_buffer, false);
|
|
|
|
if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
|
|
capture_audio_buffer->MergeFrequencyBands();
|
|
}
|
|
}
|
|
|
|
void SetupComponent(int sample_rate_hz,
|
|
GainControl::Mode mode,
|
|
int target_level_dbfs,
|
|
int stream_analog_level,
|
|
int compression_gain_db,
|
|
bool enable_limiter,
|
|
int analog_level_min,
|
|
int analog_level_max,
|
|
GainControlImpl* gain_controller) {
|
|
gain_controller->Initialize(1, sample_rate_hz);
|
|
GainControl* gc = static_cast<GainControl*>(gain_controller);
|
|
gc->Enable(true);
|
|
gc->set_mode(mode);
|
|
gc->set_stream_analog_level(stream_analog_level);
|
|
gc->set_target_level_dbfs(target_level_dbfs);
|
|
gc->set_compression_gain_db(compression_gain_db);
|
|
gc->enable_limiter(enable_limiter);
|
|
gc->set_analog_level_limits(analog_level_min, analog_level_max);
|
|
}
|
|
|
|
void RunBitExactnessTest(int sample_rate_hz,
|
|
size_t num_channels,
|
|
GainControl::Mode mode,
|
|
int target_level_dbfs,
|
|
int stream_analog_level,
|
|
int compression_gain_db,
|
|
bool enable_limiter,
|
|
int analog_level_min,
|
|
int analog_level_max,
|
|
int achieved_stream_analog_level_reference,
|
|
rtc::ArrayView<const float> output_reference) {
|
|
GainControlImpl gain_controller;
|
|
SetupComponent(sample_rate_hz, mode, target_level_dbfs, stream_analog_level,
|
|
compression_gain_db, enable_limiter, analog_level_min,
|
|
analog_level_max, &gain_controller);
|
|
|
|
const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
|
|
const StreamConfig render_config(sample_rate_hz, num_channels, false);
|
|
AudioBuffer render_buffer(
|
|
render_config.sample_rate_hz(), render_config.num_channels(),
|
|
render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
|
|
test::InputAudioFile render_file(
|
|
test::GetApmRenderTestVectorFileName(sample_rate_hz));
|
|
std::vector<float> render_input(samples_per_channel * num_channels);
|
|
|
|
const StreamConfig capture_config(sample_rate_hz, num_channels, false);
|
|
AudioBuffer capture_buffer(
|
|
capture_config.sample_rate_hz(), capture_config.num_channels(),
|
|
capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
|
|
test::InputAudioFile capture_file(
|
|
test::GetApmCaptureTestVectorFileName(sample_rate_hz));
|
|
std::vector<float> capture_input(samples_per_channel * num_channels);
|
|
|
|
for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
|
|
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
|
|
&render_file, render_input);
|
|
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
|
|
&capture_file, capture_input);
|
|
|
|
test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
|
|
test::CopyVectorToAudioBuffer(capture_config, capture_input,
|
|
&capture_buffer);
|
|
|
|
ProcessOneFrame(sample_rate_hz, &render_buffer, &capture_buffer,
|
|
&gain_controller);
|
|
}
|
|
|
|
// Extract and verify the test results.
|
|
std::vector<float> capture_output;
|
|
test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
|
|
&capture_output);
|
|
|
|
EXPECT_EQ(achieved_stream_analog_level_reference,
|
|
gain_controller.stream_analog_level());
|
|
|
|
// Compare the output with the reference. Only the first values of the output
|
|
// from last frame processed are compared in order not having to specify all
|
|
// preceeding frames as testvectors. As the algorithm being tested has a
|
|
// memory, testing only the last frame implicitly also tests the preceeding
|
|
// frames.
|
|
const float kElementErrorBound = 1.0f / 32768.0f;
|
|
EXPECT_TRUE(test::VerifyDeinterleavedArray(
|
|
capture_config.num_frames(), capture_config.num_channels(),
|
|
output_reference, capture_output, kElementErrorBound));
|
|
}
|
|
|
|
} // namespace
|
|
|
|
// TODO(peah): Activate all these tests for ARM and ARM64 once the issue on the
|
|
// Chromium ARM and ARM64 boths have been identified. This is tracked in the
|
|
// issue https://bugs.chromium.org/p/webrtc/issues/detail?id=5711.
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono8kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono8kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.006622f, -0.002747f, 0.001587f};
|
|
RunBitExactnessTest(8000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.027313f, -0.015900f, -0.028107f,
|
|
-0.027313f, -0.015900f, -0.028107f};
|
|
RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono8kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono8kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.004028f, -0.001678f, 0.000946f};
|
|
RunBitExactnessTest(8000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.003967f, -0.002777f, -0.001770f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.015411f, -0.008972f, -0.015839f,
|
|
-0.015411f, -0.008972f, -0.015839f};
|
|
RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.006104f, -0.005524f, -0.004974f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.006104f, -0.005524f, -0.004974f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono8kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono8kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.011871f, -0.004944f, 0.002838f};
|
|
RunBitExactnessTest(8000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.011749f, -0.008270f, -0.005219f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.048950f, -0.028503f, -0.050354f,
|
|
-0.048950f, -0.028503f, -0.050354f};
|
|
RunBitExactnessTest(16000, 2, GainControl::Mode::kFixedDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.018188f, -0.016418f, -0.014862f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 50;
|
|
const float kOutputReference[] = {-0.018188f, -0.016418f, -0.014862f};
|
|
RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 12;
|
|
const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 10, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 100;
|
|
const float kOutputReference[] = {-0.003998f, -0.002808f, -0.001770f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 100, 5,
|
|
true, 70, 80, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 100;
|
|
const float kOutputReference[] = {-0.004028f, -0.002838f, -0.001770f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100, 5,
|
|
false, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 100;
|
|
const float kOutputReference[] = {-0.008728f, -0.006134f, -0.003845f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 40, 100, 5,
|
|
true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
|
|
defined(WEBRTC_ANDROID))
|
|
TEST(GainControlBitExactnessTest,
|
|
Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
|
|
#else
|
|
TEST(GainControlBitExactnessTest,
|
|
DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
|
|
#endif
|
|
const int kStreamAnalogLevelReference = 100;
|
|
const float kOutputReference[] = {-0.005859f, -0.004120f, -0.002594f};
|
|
RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100,
|
|
30, true, 0, 100, kStreamAnalogLevelReference,
|
|
kOutputReference);
|
|
}
|
|
|
|
} // namespace webrtc
|