Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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