diff options
Diffstat (limited to 'third_party/libwebrtc/api/audio/test')
4 files changed, 305 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio/test/BUILD.gn b/third_party/libwebrtc/api/audio/test/BUILD.gn new file mode 100644 index 0000000000..dfe8c32f80 --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/BUILD.gn @@ -0,0 +1,30 @@ +# Copyright (c) 2018 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. + +import("../../../webrtc.gni") +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} + +if (rtc_include_tests) { + rtc_library("audio_api_unittests") { + testonly = true + sources = [ + "audio_frame_unittest.cc", + "echo_canceller3_config_json_unittest.cc", + "echo_canceller3_config_unittest.cc", + ] + deps = [ + "..:aec3_config", + "..:aec3_config_json", + "..:audio_frame_api", + "../../../test:test_support", + ] + } +} diff --git a/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc b/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc new file mode 100644 index 0000000000..dbf45ceabc --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc @@ -0,0 +1,136 @@ +/* + * Copyright 2018 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 "api/audio/audio_frame.h" + +#include <stdint.h> +#include <string.h> // memcmp + +#include "test/gtest.h" + +namespace webrtc { + +namespace { + +bool AllSamplesAre(int16_t sample, const AudioFrame& frame) { + const int16_t* frame_data = frame.data(); + for (size_t i = 0; i < frame.max_16bit_samples(); i++) { + if (frame_data[i] != sample) { + return false; + } + } + return true; +} + +constexpr uint32_t kTimestamp = 27; +constexpr int kSampleRateHz = 16000; +constexpr size_t kNumChannelsMono = 1; +constexpr size_t kNumChannelsStereo = 2; +constexpr size_t kNumChannels5_1 = 6; +constexpr size_t kSamplesPerChannel = kSampleRateHz / 100; + +} // namespace + +TEST(AudioFrameTest, FrameStartsMuted) { + AudioFrame frame; + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) { + AudioFrame frame; + frame.mutable_data(); + EXPECT_FALSE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, MutedFrameBufferIsZeroed) { + AudioFrame frame; + int16_t* frame_data = frame.mutable_data(); + for (size_t i = 0; i < frame.max_16bit_samples(); i++) { + frame_data[i] = 17; + } + ASSERT_TRUE(AllSamplesAre(17, frame)); + frame.Mute(); + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UpdateFrameMono) { + AudioFrame frame; + int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17}; + frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz, + AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannelsMono); + + EXPECT_EQ(kTimestamp, frame.timestamp_); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz()); + EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_); + EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_); + EXPECT_EQ(kNumChannelsMono, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame.channel_layout()); + + EXPECT_FALSE(frame.muted()); + EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples))); + + frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UpdateFrameMultiChannel) { + AudioFrame frame; + frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsStereo); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kNumChannelsStereo, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame.channel_layout()); + + frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannels5_1); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kNumChannels5_1, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame.channel_layout()); +} + +TEST(AudioFrameTest, CopyFrom) { + AudioFrame frame1; + AudioFrame frame2; + + int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17}; + frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz, + AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + frame1.CopyFrom(frame2); + + EXPECT_EQ(frame2.timestamp_, frame1.timestamp_); + EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_); + EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_); + EXPECT_EQ(frame2.speech_type_, frame1.speech_type_); + EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_); + EXPECT_EQ(frame2.num_channels_, frame1.num_channels_); + + EXPECT_EQ(frame2.muted(), frame1.muted()); + EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples))); + + frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + frame1.CopyFrom(frame2); + + EXPECT_EQ(frame2.muted(), frame1.muted()); + EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples))); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/test/echo_canceller3_config_json_unittest.cc b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_json_unittest.cc new file mode 100644 index 0000000000..4146dda9fe --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_json_unittest.cc @@ -0,0 +1,93 @@ +/* + * Copyright 2018 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 "api/audio/echo_canceller3_config_json.h" + +#include "api/audio/echo_canceller3_config.h" +#include "test/gtest.h" + +namespace webrtc { + +TEST(EchoCanceller3JsonHelpers, ToStringAndParseJson) { + EchoCanceller3Config cfg; + cfg.delay.down_sampling_factor = 1u; + cfg.delay.log_warning_on_delay_changes = true; + cfg.filter.refined.error_floor = 2.f; + cfg.filter.coarse_initial.length_blocks = 3u; + cfg.filter.high_pass_filter_echo_reference = + !cfg.filter.high_pass_filter_echo_reference; + cfg.comfort_noise.noise_floor_dbfs = 100.f; + cfg.echo_model.model_reverb_in_nonlinear_mode = false; + cfg.suppressor.normal_tuning.mask_hf.enr_suppress = .5f; + cfg.suppressor.subband_nearend_detection.nearend_average_blocks = 3; + cfg.suppressor.subband_nearend_detection.subband1 = {1, 3}; + cfg.suppressor.subband_nearend_detection.subband1 = {4, 5}; + cfg.suppressor.subband_nearend_detection.nearend_threshold = 2.f; + cfg.suppressor.subband_nearend_detection.snr_threshold = 100.f; + cfg.multi_channel.detect_stereo_content = + !cfg.multi_channel.detect_stereo_content; + cfg.multi_channel.stereo_detection_threshold += 1.0f; + cfg.multi_channel.stereo_detection_timeout_threshold_seconds += 1; + cfg.multi_channel.stereo_detection_hysteresis_seconds += 1; + std::string json_string = Aec3ConfigToJsonString(cfg); + EchoCanceller3Config cfg_transformed = Aec3ConfigFromJsonString(json_string); + + // Expect unchanged values to remain default. + EXPECT_EQ(cfg.ep_strength.default_len, + cfg_transformed.ep_strength.default_len); + EXPECT_EQ(cfg.ep_strength.nearend_len, + cfg_transformed.ep_strength.nearend_len); + EXPECT_EQ(cfg.suppressor.normal_tuning.mask_lf.enr_suppress, + cfg_transformed.suppressor.normal_tuning.mask_lf.enr_suppress); + + // Expect changed values to carry through the transformation. + EXPECT_EQ(cfg.delay.down_sampling_factor, + cfg_transformed.delay.down_sampling_factor); + EXPECT_EQ(cfg.delay.log_warning_on_delay_changes, + cfg_transformed.delay.log_warning_on_delay_changes); + EXPECT_EQ(cfg.filter.coarse_initial.length_blocks, + cfg_transformed.filter.coarse_initial.length_blocks); + EXPECT_EQ(cfg.filter.refined.error_floor, + cfg_transformed.filter.refined.error_floor); + EXPECT_EQ(cfg.filter.high_pass_filter_echo_reference, + cfg_transformed.filter.high_pass_filter_echo_reference); + EXPECT_EQ(cfg.comfort_noise.noise_floor_dbfs, + cfg_transformed.comfort_noise.noise_floor_dbfs); + EXPECT_EQ(cfg.echo_model.model_reverb_in_nonlinear_mode, + cfg_transformed.echo_model.model_reverb_in_nonlinear_mode); + EXPECT_EQ(cfg.suppressor.normal_tuning.mask_hf.enr_suppress, + cfg_transformed.suppressor.normal_tuning.mask_hf.enr_suppress); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.nearend_average_blocks, + cfg_transformed.suppressor.subband_nearend_detection + .nearend_average_blocks); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.subband1.low, + cfg_transformed.suppressor.subband_nearend_detection.subband1.low); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.subband1.high, + cfg_transformed.suppressor.subband_nearend_detection.subband1.high); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.subband2.low, + cfg_transformed.suppressor.subband_nearend_detection.subband2.low); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.subband2.high, + cfg_transformed.suppressor.subband_nearend_detection.subband2.high); + EXPECT_EQ( + cfg.suppressor.subband_nearend_detection.nearend_threshold, + cfg_transformed.suppressor.subband_nearend_detection.nearend_threshold); + EXPECT_EQ(cfg.suppressor.subband_nearend_detection.snr_threshold, + cfg_transformed.suppressor.subband_nearend_detection.snr_threshold); + EXPECT_EQ(cfg.multi_channel.detect_stereo_content, + cfg_transformed.multi_channel.detect_stereo_content); + EXPECT_EQ(cfg.multi_channel.stereo_detection_threshold, + cfg_transformed.multi_channel.stereo_detection_threshold); + EXPECT_EQ( + cfg.multi_channel.stereo_detection_timeout_threshold_seconds, + cfg_transformed.multi_channel.stereo_detection_timeout_threshold_seconds); + EXPECT_EQ(cfg.multi_channel.stereo_detection_hysteresis_seconds, + cfg_transformed.multi_channel.stereo_detection_hysteresis_seconds); +} +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc new file mode 100644 index 0000000000..91312a0f40 --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc @@ -0,0 +1,46 @@ +/* + * Copyright 2018 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 "api/audio/echo_canceller3_config.h" + +#include "api/audio/echo_canceller3_config_json.h" +#include "test/gtest.h" + +namespace webrtc { + +TEST(EchoCanceller3Config, ValidConfigIsNotModified) { + EchoCanceller3Config config; + EXPECT_TRUE(EchoCanceller3Config::Validate(&config)); + EchoCanceller3Config default_config; + EXPECT_EQ(Aec3ConfigToJsonString(config), + Aec3ConfigToJsonString(default_config)); +} + +TEST(EchoCanceller3Config, InvalidConfigIsCorrected) { + // Change a parameter and validate. + EchoCanceller3Config config; + config.echo_model.min_noise_floor_power = -1600000.f; + EXPECT_FALSE(EchoCanceller3Config::Validate(&config)); + EXPECT_GE(config.echo_model.min_noise_floor_power, 0.f); + // Verify remaining parameters are unchanged. + EchoCanceller3Config default_config; + config.echo_model.min_noise_floor_power = + default_config.echo_model.min_noise_floor_power; + EXPECT_EQ(Aec3ConfigToJsonString(config), + Aec3ConfigToJsonString(default_config)); +} + +TEST(EchoCanceller3Config, ValidatedConfigsAreValid) { + EchoCanceller3Config config; + config.delay.down_sampling_factor = 983; + EXPECT_FALSE(EchoCanceller3Config::Validate(&config)); + EXPECT_TRUE(EchoCanceller3Config::Validate(&config)); +} +} // namespace webrtc |