InputVolumeController: Remove two unit test helper functions
Remove deprecated unit test helper functions CallPreProcessAudioBuffer() and CallPreProcForChangingAudio(). Replace the use of these functions with CallAgcSequence(). Remove a duplicate unit test using one of these functions. The new calls follow the API contract. Bug: webrtc:7494 Change-Id: Idc033cb48f4fab1814c4c6e0f23edc4a6a9faa64 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285960 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38826}
This commit is contained in:
Hanna Silen
WebRTC LUCI CQ
parent
38a6002548
commit
fcbf3724eb
@ -81,45 +81,6 @@ std::unique_ptr<InputVolumeController> CreateInputVolumeController(
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config);
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}
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// Deprecated.
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// TODO(bugs.webrtc.org/7494): Delete this helper, use
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// `InputVolumeControllerTestHelper::CallAgcSequence()` instead.
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// Calls `AnalyzePreProcess()` on `controller` `num_calls` times. `peak_ratio`
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// is a value in [0, 1] which determines the amplitude of the samples (1 maps to
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// full scale). The first half of the calls is made on frames which are half
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// filled with zeros in order to simulate a signal with different crest factors.
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void CallPreProcessAudioBuffer(int num_calls,
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float peak_ratio,
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InputVolumeController& controller) {
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RTC_DCHECK_LE(peak_ratio, 1.0f);
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AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
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kNumChannels, kSampleRateHz, kNumChannels);
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const int num_channels = audio_buffer.num_channels();
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const int num_frames = audio_buffer.num_frames();
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// Make half of the calls with half zeroed frames.
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for (int ch = 0; ch < num_channels; ++ch) {
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// 50% of the samples in one frame are zero.
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for (int i = 0; i < num_frames; i += 2) {
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audio_buffer.channels()[ch][i] = peak_ratio * 32767.0f;
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audio_buffer.channels()[ch][i + 1] = 0.0f;
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}
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}
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for (int n = 0; n < num_calls / 2; ++n) {
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controller.AnalyzePreProcess(audio_buffer);
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}
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// Make the remaining half of the calls with frames whose samples are all set.
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for (int ch = 0; ch < num_channels; ++ch) {
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for (int i = 0; i < num_frames; ++i) {
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audio_buffer.channels()[ch][i] = peak_ratio * 32767.0f;
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}
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}
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for (int n = 0; n < num_calls - num_calls / 2; ++n) {
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controller.AnalyzePreProcess(audio_buffer);
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}
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}
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constexpr char kMinInputVolumeFieldTrial[] = "WebRTC-Audio-Agc2-MinInputVolume";
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std::string GetAgcMinInputVolumeFieldTrial(const std::string& value) {
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@ -172,17 +133,33 @@ void WriteAudioBufferSamples(float samples_value,
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}
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}
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// (Over)writes samples in `audio_buffer`. Alternates samples `samples_value`
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// and zero.
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void WriteAlternatingAudioBufferSamples(float samples_value,
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AudioBuffer& audio_buffer) {
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RTC_DCHECK_GE(samples_value, kMinSample);
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RTC_DCHECK_LE(samples_value, kMaxSample);
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const int num_channels = audio_buffer.num_channels();
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const int num_frames = audio_buffer.num_frames();
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for (int ch = 0; ch < num_channels; ++ch) {
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for (int i = 0; i < num_frames; i += 2) {
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audio_buffer.channels()[ch][i] = samples_value;
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audio_buffer.channels()[ch][i + 1] = 0.0f;
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}
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}
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}
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// Deprecated.
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// TODO(bugs.webrtc.org/7494): Delete this helper, use
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// `InputVolumeControllerTestHelper::CallAgcSequence()` instead.
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void CallPreProcessAndProcess(int num_calls,
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const AudioBuffer& audio_buffer,
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float speech_probability,
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absl::optional<float> speech_level,
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absl::optional<float> speech_level_dbfs,
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InputVolumeController& controller) {
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for (int n = 0; n < num_calls; ++n) {
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controller.AnalyzePreProcess(audio_buffer);
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controller.Process(speech_probability, speech_level);
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controller.Process(speech_probability, speech_level_dbfs);
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}
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}
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@ -214,12 +191,12 @@ class SpeechSamplesReader {
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// `gain_db` and feeds the frames into `controller` by calling
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// `AnalyzePreProcess()` and `Process()` for each frame. Reads the number of
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// 10 ms frames available in the PCM file if `num_frames` is too large - i.e.,
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// does not loop. `speech_probability` and `speech_level` are passed to
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// does not loop. `speech_probability` and `speech_level_dbfs` are passed to
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// `Process()`.
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void Feed(int num_frames,
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int gain_db,
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float speech_probability,
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absl::optional<float> speech_level,
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absl::optional<float> speech_level_dbfs,
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InputVolumeController& controller) {
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float gain = std::pow(10.0f, gain_db / 20.0f); // From dB to linear gain.
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is_.seekg(0, is_.beg); // Start from the beginning of the PCM file.
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@ -239,7 +216,7 @@ class SpeechSamplesReader {
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});
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controller.AnalyzePreProcess(audio_buffer_);
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controller.Process(speech_probability, speech_level);
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controller.Process(speech_probability, speech_level_dbfs);
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}
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}
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@ -289,15 +266,16 @@ constexpr InputVolumeControllerConfig GetInputVolumeControllerTestConfig() {
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class InputVolumeControllerTestHelper {
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public:
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// Ctor. Initializes `audio_buffer` with zeros.
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InputVolumeControllerTestHelper()
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// TODO(bugs.webrtc.org/7494): Remove the default argument.
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InputVolumeControllerTestHelper(const InputVolumeController::Config& config =
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GetInputVolumeControllerTestConfig())
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: audio_buffer(kSampleRateHz,
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kNumChannels,
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kSampleRateHz,
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kNumChannels,
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kSampleRateHz,
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kNumChannels),
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controller(/*num_capture_channels=*/1,
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GetInputVolumeControllerTestConfig()) {
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controller(/*num_capture_channels=*/1, config) {
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controller.Initialize();
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WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
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audio_buffer);
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@ -310,12 +288,18 @@ class InputVolumeControllerTestHelper {
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// Returns the recommended input volume.
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int CallAgcSequence(int applied_input_volume,
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float speech_probability,
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absl::optional<float> speech_level) {
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controller.set_stream_analog_level(applied_input_volume);
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controller.AnalyzePreProcess(audio_buffer);
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controller.Process(speech_probability, speech_level);
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absl::optional<float> speech_level_dbfs,
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int num_calls = 1) {
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RTC_DCHECK_GE(num_calls, 1);
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int volume = applied_input_volume;
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return controller.recommended_analog_level();
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for (int i = 0; i < num_calls; ++i) {
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controller.set_stream_analog_level(volume);
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controller.AnalyzePreProcess(audio_buffer);
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controller.Process(speech_probability, speech_level_dbfs);
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volume = controller.recommended_analog_level();
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}
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return volume;
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}
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// Deprecated.
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@ -323,9 +307,9 @@ class InputVolumeControllerTestHelper {
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// `CallAgcSequence()`.
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void CallProcess(int num_calls,
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float speech_probability,
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absl::optional<float> speech_level) {
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absl::optional<float> speech_level_dbfs) {
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for (int i = 0; i < num_calls; ++i) {
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controller.Process(speech_probability, speech_level);
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controller.Process(speech_probability, speech_level_dbfs);
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}
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}
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@ -342,36 +326,6 @@ class InputVolumeControllerTestHelper {
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}
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}
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// Deprecated.
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// TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
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// `CallAgcSequence()`.
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void CallPreProcForChangingAudio(int num_calls, float peak_ratio) {
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RTC_DCHECK_GE(peak_ratio, 0.0f);
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RTC_DCHECK_LE(peak_ratio, 1.0f);
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const float samples_value = peak_ratio * 32767.0f;
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// Make half of the calls on a frame where the samples alternate
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// `sample_values` and zeros.
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WriteAudioBufferSamples(samples_value, /*clipped_ratio=*/0.0f,
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audio_buffer);
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for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
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for (size_t k = 1; k < audio_buffer.num_frames(); k += 2) {
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audio_buffer.channels()[ch][k] = 0.0f;
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}
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}
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for (int i = 0; i < num_calls / 2; ++i) {
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controller.AnalyzePreProcess(audio_buffer);
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}
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// Make half of thecalls on a frame where all the samples equal
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// `sample_values`.
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WriteAudioBufferSamples(samples_value, /*clipped_ratio=*/0.0f,
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audio_buffer);
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for (int i = 0; i < num_calls - num_calls / 2; ++i) {
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controller.AnalyzePreProcess(audio_buffer);
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}
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}
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AudioBuffer audio_buffer;
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InputVolumeController controller;
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};
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@ -745,28 +699,23 @@ TEST_P(InputVolumeControllerParametrizedTest, TakesNoActionOnZeroMicVolume) {
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TEST_P(InputVolumeControllerParametrizedTest, ClippingDetectionLowersVolume) {
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InputVolumeControllerTestHelper helper;
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helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
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kSpeechLevel);
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int volume = helper.CallAgcSequence(/*applied_input_volume=*/255,
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kHighSpeechProbability, kSpeechLevel,
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/*num_calls=*/1);
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EXPECT_EQ(helper.controller.recommended_analog_level(), 255);
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helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
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EXPECT_EQ(helper.controller.recommended_analog_level(), 255);
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helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/1.0f);
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EXPECT_EQ(helper.controller.recommended_analog_level(), 240);
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}
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EXPECT_EQ(volume, 255);
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TEST_P(InputVolumeControllerParametrizedTest,
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DisabledClippingPredictorDoesNotLowerVolume) {
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InputVolumeControllerTestHelper helper;
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helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
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kSpeechLevel);
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WriteAlternatingAudioBufferSamples(0.99f * kMaxSample, helper.audio_buffer);
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volume = helper.CallAgcSequence(volume, kHighSpeechProbability, kSpeechLevel,
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/*num_calls=*/100);
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EXPECT_FALSE(helper.controller.clipping_predictor_enabled());
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EXPECT_EQ(helper.controller.recommended_analog_level(), 255);
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helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
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EXPECT_EQ(helper.controller.recommended_analog_level(), 255);
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helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
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EXPECT_EQ(helper.controller.recommended_analog_level(), 255);
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EXPECT_EQ(volume, 255);
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WriteAlternatingAudioBufferSamples(kMaxSample, helper.audio_buffer);
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volume = helper.CallAgcSequence(volume, kHighSpeechProbability, kSpeechLevel,
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/*num_calls=*/100);
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EXPECT_EQ(volume, 240);
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}
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TEST(InputVolumeControllerTest, MinInputVolumeDefault) {
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@ -866,10 +815,10 @@ TEST(InputVolumeControllerTest, MinInputVolumeCheckMinLevelWithClipping) {
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// adjustment of the analog gain. Use low speech probability to limit the
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// volume changes to clipping handling.
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CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
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kLowSpeechProbability, /*speech_level=*/-42.0f,
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kLowSpeechProbability, /*speech_level_dbfs=*/-42.0f,
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*controller);
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CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
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kLowSpeechProbability, /*speech_level=*/-42.0f,
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kLowSpeechProbability, /*speech_level_dbfs=*/-42.0f,
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*manager_with_override);
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// Make sure that an adaptation occurred.
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@ -922,10 +871,10 @@ TEST(InputVolumeControllerTest,
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// adjustment of the analog gain.
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CallPreProcessAndProcess(
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/*num_calls=*/400, audio_buffer, kHighSpeechProbability,
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/*speech_level=*/-18.0f, *controller);
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/*speech_level_dbfs=*/-18.0f, *controller);
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CallPreProcessAndProcess(
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/*num_calls=*/400, audio_buffer, kHighSpeechProbability,
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/*speech_level=*/-18.0f, *manager_with_override);
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/*speech_level_dbfs=*/-18.0f, *manager_with_override);
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// Make sure that an adaptation occurred.
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ASSERT_GT(controller->recommended_analog_level(), 0);
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@ -981,10 +930,10 @@ TEST(InputVolumeControllerTest, MinInputVolumeCompareMicLevelWithClipping) {
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// adjustment of the analog gain. Use low speech probability to limit the
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// volume changes to clipping handling.
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CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
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kLowSpeechProbability, /*speech_level=*/-18,
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kLowSpeechProbability, /*speech_level_dbfs=*/-18,
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*controller);
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CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
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kLowSpeechProbability, /*speech_level=*/-18,
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kLowSpeechProbability, /*speech_level_dbfs=*/-18,
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*manager_with_override);
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// Make sure that an adaptation occurred.
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@ -1045,11 +994,11 @@ TEST(InputVolumeControllerTest,
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CallPreProcessAndProcess(
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/*num_calls=*/400, audio_buffer,
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/*speech_probability=*/0.7f,
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/*speech_level=*/-18.0f, *controller);
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/*speech_level_dbfs=*/-18.0f, *controller);
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CallPreProcessAndProcess(
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/*num_calls=*/400, audio_buffer,
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/*speech_probability=*/0.7f,
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/*speech_level=*/-18.0f, *manager_with_override);
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/*speech_level_dbfs=*/-18.0f, *manager_with_override);
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// Make sure that an adaptation occurred.
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ASSERT_GT(controller->recommended_analog_level(), 0);
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@ -1112,134 +1061,187 @@ TEST_P(InputVolumeControllerParametrizedTest,
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TEST_P(InputVolumeControllerParametrizedTest,
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DisableClippingPredictorDoesNotLowerVolume) {
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AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
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kNumChannels, kSampleRateHz, kNumChannels);
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int volume = 255;
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InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
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config.enable_clipping_predictor = false;
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InputVolumeController controller(/*num_capture_channels=*/1, config);
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controller.Initialize();
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controller.set_stream_analog_level(/*level=*/255);
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EXPECT_FALSE(controller.clipping_predictor_enabled());
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EXPECT_FALSE(controller.use_clipping_predictor_step());
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EXPECT_EQ(controller.recommended_analog_level(), 255);
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controller.Process(kHighSpeechProbability, kSpeechLevel);
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CallPreProcessAudioBuffer(/*num_calls=*/10, /*peak_ratio=*/0.99f, controller);
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EXPECT_EQ(controller.recommended_analog_level(), 255);
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CallPreProcessAudioBuffer(/*num_calls=*/300, /*peak_ratio=*/0.99f,
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controller);
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EXPECT_EQ(controller.recommended_analog_level(), 255);
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CallPreProcessAudioBuffer(/*num_calls=*/10, /*peak_ratio=*/0.99f, controller);
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EXPECT_EQ(controller.recommended_analog_level(), 255);
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auto helper = InputVolumeControllerTestHelper(config);
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helper.controller.Initialize();
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EXPECT_FALSE(helper.controller.clipping_predictor_enabled());
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EXPECT_FALSE(helper.controller.use_clipping_predictor_step());
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// Expect no change if clipping prediction is enabled.
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for (int j = 0; j < 31; ++j) {
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WriteAlternatingAudioBufferSamples(0.99f * kMaxSample, helper.audio_buffer);
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volume = helper.CallAgcSequence(volume, kLowSpeechProbability, kSpeechLevel,
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/*num_calls=*/5);
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WriteAudioBufferSamples(0.99f * kMaxSample, /*clipped_ratio=*/0.0f,
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helper.audio_buffer);
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volume = helper.CallAgcSequence(volume, kLowSpeechProbability, kSpeechLevel,
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/*num_calls=*/5);
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EXPECT_EQ(volume, 255);
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}
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}
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// TODO(bugs.webrtc.org/7494): Split into several smaller tests.
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TEST_P(InputVolumeControllerParametrizedTest,
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UsedClippingPredictionsProduceLowerAnalogLevels) {
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AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
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kNumChannels, kSampleRateHz, kNumChannels);
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InputVolumeControllerConfig config_with_prediction =
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GetInputVolumeControllerTestConfig();
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config_with_prediction.enable_clipping_predictor = true;
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InputVolumeControllerConfig config_without_prediction =
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GetInputVolumeControllerTestConfig();
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config_without_prediction.enable_clipping_predictor = false;
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InputVolumeController manager_without_prediction(/*num_capture_channels=*/1,
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config_without_prediction);
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InputVolumeController manager_with_prediction(/*num_capture_channels=*/1,
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config_with_prediction);
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manager_with_prediction.Initialize();
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manager_without_prediction.Initialize();
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constexpr int kInitialLevel = 255;
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constexpr float kClippingPeakRatio = 1.0f;
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constexpr float kCloseToClippingPeakRatio = 0.99f;
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constexpr float kZeroPeakRatio = 0.0f;
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manager_with_prediction.set_stream_analog_level(kInitialLevel);
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manager_without_prediction.set_stream_analog_level(kInitialLevel);
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int volume_1 = kInitialLevel;
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int volume_2 = kInitialLevel;
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manager_with_prediction.Process(kHighSpeechProbability, kSpeechLevel);
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manager_without_prediction.Process(kHighSpeechProbability, kSpeechLevel);
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// Create two helpers, one with clipping prediction and one without.
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auto config_1 = GetInputVolumeControllerTestConfig();
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auto config_2 = GetInputVolumeControllerTestConfig();
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config_1.enable_clipping_predictor = true;
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config_2.enable_clipping_predictor = false;
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auto helper_1 = InputVolumeControllerTestHelper(config_1);
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auto helper_2 = InputVolumeControllerTestHelper(config_2);
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helper_1.controller.Initialize();
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helper_2.controller.Initialize();
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EXPECT_TRUE(manager_with_prediction.clipping_predictor_enabled());
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EXPECT_FALSE(manager_without_prediction.clipping_predictor_enabled());
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EXPECT_TRUE(manager_with_prediction.use_clipping_predictor_step());
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EXPECT_EQ(manager_with_prediction.recommended_analog_level(), kInitialLevel);
|
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EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel);
|
||||
EXPECT_TRUE(helper_1.controller.clipping_predictor_enabled());
|
||||
EXPECT_FALSE(helper_2.controller.clipping_predictor_enabled());
|
||||
EXPECT_TRUE(helper_1.controller.use_clipping_predictor_step());
|
||||
|
||||
// Expect a change in the analog level when the prediction step is used.
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel);
|
||||
// Expect a change if clipping prediction is enabled.
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_1.audio_buffer);
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel);
|
||||
|
||||
// Expect no change during waiting.
|
||||
CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel);
|
||||
for (int i = 0; i < kClippedWaitFrames / 10; ++i) {
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_1.audio_buffer);
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel);
|
||||
}
|
||||
|
||||
// Expect a change when the prediction step is used.
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 2 * kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel);
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_1.audio_buffer);
|
||||
WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
|
||||
/*clipped_ratio=*/0.0f, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - 2 * kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel);
|
||||
|
||||
// Expect no change when clipping is not detected or predicted.
|
||||
CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 2 * kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel);
|
||||
for (int i = 0; i < 2 * kClippedWaitFrames / 10; ++i) {
|
||||
WriteAlternatingAudioBufferSamples(/*samples_value=*/0.0f,
|
||||
helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(/*samples_value=*/0.0f,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
|
||||
helper_1.audio_buffer);
|
||||
WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
}
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - 2 * kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel);
|
||||
|
||||
// Expect a change for clipping frames.
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 3 * kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel - kClippedLevelStep);
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 1);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 1);
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - 3 * kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel - kClippedLevelStep);
|
||||
|
||||
// Expect no change during waiting.
|
||||
CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 3 * kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel - kClippedLevelStep);
|
||||
for (int i = 0; i < kClippedWaitFrames / 10; ++i) {
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
|
||||
WriteAudioBufferSamples(kMaxSample, /*clipped_ratio=*/1.0f,
|
||||
helper_1.audio_buffer);
|
||||
WriteAudioBufferSamples(kMaxSample, /*clipped_ratio=*/1.0f,
|
||||
helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 5);
|
||||
}
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - 3 * kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel - kClippedLevelStep);
|
||||
|
||||
// Expect a change for clipping frames.
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
|
||||
manager_with_prediction);
|
||||
CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
|
||||
manager_without_prediction);
|
||||
EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 4 * kClippedLevelStep);
|
||||
EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
|
||||
kInitialLevel - 2 * kClippedLevelStep);
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
|
||||
WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
|
||||
volume_1 = helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
|
||||
kSpeechLevel, 1);
|
||||
volume_2 = helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
|
||||
kSpeechLevel, 1);
|
||||
|
||||
EXPECT_EQ(volume_1, kInitialLevel - 4 * kClippedLevelStep);
|
||||
EXPECT_EQ(volume_2, kInitialLevel - 2 * kClippedLevelStep);
|
||||
}
|
||||
|
||||
// Checks that passing an empty speech level has no effect on the input volume.
|
||||
@ -1285,18 +1287,18 @@ TEST(InputVolumeControllerTest, UpdateInputVolumeWaitFramesIsEffective) {
|
||||
SpeechSamplesReader reader_1;
|
||||
SpeechSamplesReader reader_2;
|
||||
reader_1.Feed(/*num_frames=*/99, /*gain_db=*/0, kHighSpeechProbability,
|
||||
/*speech_level=*/-42.0f, *controller_wait_0);
|
||||
/*speech_level_dbfs=*/-42.0f, *controller_wait_0);
|
||||
reader_2.Feed(/*num_frames=*/99, /*gain_db=*/0, kHighSpeechProbability,
|
||||
/*speech_level=*/-42.0f, *controller_wait_100);
|
||||
/*speech_level_dbfs=*/-42.0f, *controller_wait_100);
|
||||
|
||||
// Check that adaptation only occurs if enough frames have been processed.
|
||||
ASSERT_GT(controller_wait_0->recommended_analog_level(), kInputVolume);
|
||||
ASSERT_EQ(controller_wait_100->recommended_analog_level(), kInputVolume);
|
||||
|
||||
reader_1.Feed(/*num_frames=*/1, /*gain_db=*/0, kHighSpeechProbability,
|
||||
/*speech_level=*/-42.0f, *controller_wait_0);
|
||||
/*speech_level_dbfs=*/-42.0f, *controller_wait_0);
|
||||
reader_2.Feed(/*num_frames=*/1, /*gain_db=*/0, kHighSpeechProbability,
|
||||
/*speech_level=*/-42.0f, *controller_wait_100);
|
||||
/*speech_level_dbfs=*/-42.0f, *controller_wait_100);
|
||||
|
||||
// Check that adaptation only occurs when enough frames have been processed.
|
||||
ASSERT_GT(controller_wait_0->recommended_analog_level(), kInputVolume);
|
||||
@ -1326,30 +1328,30 @@ TEST(InputVolumeControllerTest, SpeechRatioThresholdIsEffective) {
|
||||
SpeechSamplesReader reader_2;
|
||||
|
||||
reader_1.Feed(/*num_frames=*/1, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.7f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.7f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/1, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.4f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.4f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_EQ(controller_1->recommended_analog_level(), kInputVolume);
|
||||
ASSERT_EQ(controller_2->recommended_analog_level(), kInputVolume);
|
||||
|
||||
reader_1.Feed(/*num_frames=*/2, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.4f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.4f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/2, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.4f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.4f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_EQ(controller_1->recommended_analog_level(), kInputVolume);
|
||||
ASSERT_EQ(controller_2->recommended_analog_level(), kInputVolume);
|
||||
|
||||
reader_1.Feed(/*num_frames=*/7, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.7f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.7f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/7, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.7f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.7f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_GT(controller_1->recommended_analog_level(), kInputVolume);
|
||||
@ -1383,30 +1385,30 @@ TEST(InputVolumeControllerTest, SpeechProbabilityThresholdIsEffective) {
|
||||
// processsed and `reader_2` to process inputs that won't make the volume
|
||||
// to be adjusted.
|
||||
reader_1.Feed(/*num_frames=*/1, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.5f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.5f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/1, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.49f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.49f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_EQ(controller_1->recommended_analog_level(), kInputVolume);
|
||||
ASSERT_EQ(controller_2->recommended_analog_level(), kInputVolume);
|
||||
|
||||
reader_1.Feed(/*num_frames=*/2, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.49f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.49f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/2, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.49f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.49f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_EQ(controller_1->recommended_analog_level(), kInputVolume);
|
||||
ASSERT_EQ(controller_2->recommended_analog_level(), kInputVolume);
|
||||
|
||||
reader_1.Feed(/*num_frames=*/7, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.5f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.5f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_1);
|
||||
reader_2.Feed(/*num_frames=*/7, /*gain_db=*/0,
|
||||
/*speech_probability=*/0.5f, /*speech_level=*/-42.0f,
|
||||
/*speech_probability=*/0.5f, /*speech_level_dbfs=*/-42.0f,
|
||||
*controller_2);
|
||||
|
||||
ASSERT_GT(controller_1->recommended_analog_level(), kInputVolume);
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user