From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/audio/test/BUILD.gn | 29 +++++ .../api/audio/test/audio_frame_unittest.cc | 136 +++++++++++++++++++++ .../audio/test/echo_canceller3_config_unittest.cc | 46 +++++++ 3 files changed, 211 insertions(+) create mode 100644 third_party/libwebrtc/api/audio/test/BUILD.gn create mode 100644 third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc create mode 100644 third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc (limited to 'third_party/libwebrtc/api/audio/test') 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..8d5822fa54 --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/BUILD.gn @@ -0,0 +1,29 @@ +# 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_unittest.cc", + ] + deps = [ + "..:aec3_config", + "..:audio_frame_api", + "../../../modules/audio_processing:aec3_config_json", + "../../../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 +#include // 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_unittest.cc b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc new file mode 100644 index 0000000000..da0255806e --- /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 "modules/audio_processing/test/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 -- cgit v1.2.3