| /* |
| * 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->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, |
| 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, |
| 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[] = {-0.006134f, -0.005524f, -0.005005f}; |
| 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[] = {-0.006134f, -0.005524f, -0.005005}; |
| 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, |
| 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[] = {-0.048896f, -0.028479f, -0.050345f, |
| -0.048896f, -0.028479f, -0.050345f}; |
| 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[] = {-0.018158f, -0.016357f, -0.014832f}; |
| 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[] = {-0.018158f, -0.016357f, -0.014832f}; |
| 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 |
