Activate the pre-amplifier in AudioProcessing.
It's a module for applying a gain to the capture signal.
The gain is the first processing step in APM.
After this CL, these two features work:
* The PreAmplifier can be activated with
AudioProcessing::Config::pre_amplifier
* The PreApmlifier can be controlled after APM creation by
AudioProcessing::SetRuntimeSetting.
What's left is a change to AecDumps and to AecDump-replay.
NOTRY=True # 1-line change, tests just passed.
Bug: webrtc:9138
Change-Id: I85b3af511695b0a9cec2eed6fee7f05080305e1d
Reviewed-on: https://webrtc-review.googlesource.com/69811
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22881}diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index cd2bba6..2ea0159 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -33,6 +33,24 @@
MOCK_CONST_METHOD0(Release, rtc::RefCountReleaseStatus());
};
+void GenerateFixedFrame(int16_t audio_level,
+ size_t input_rate,
+ size_t num_channels,
+ AudioFrame* fixed_frame) {
+ const size_t samples_per_input_channel = rtc::CheckedDivExact(
+ input_rate, static_cast<size_t>(rtc::CheckedDivExact(
+ 1000, AudioProcessing::kChunkSizeMs)));
+ fixed_frame->samples_per_channel_ = samples_per_input_channel;
+ fixed_frame->sample_rate_hz_ = input_rate;
+ fixed_frame->num_channels_ = num_channels;
+
+ RTC_DCHECK_LE(samples_per_input_channel * num_channels,
+ AudioFrame::kMaxDataSizeSamples);
+ for (size_t i = 0; i < samples_per_input_channel * num_channels; ++i) {
+ fixed_frame->mutable_data()[i] = audio_level;
+ }
+}
+
} // namespace
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
@@ -75,9 +93,33 @@
}
TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
- // TODO(bugs.chromium.org/9138): Implement this test as soon as the pre-gain
- // sub-module is implemented and it is notified by HandleRuntimeSettings()
- // when the gain changes.
+ std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
+ webrtc::AudioProcessing::Config apm_config;
+ apm_config.pre_amplifier.enabled = true;
+ apm_config.pre_amplifier.fixed_gain_factor = 1.f;
+ apm->ApplyConfig(apm_config);
+
+ AudioFrame frame;
+ constexpr int16_t audio_level = 10000;
+ constexpr size_t input_rate = 48000;
+ constexpr size_t num_channels = 2;
+
+ GenerateFixedFrame(audio_level, input_rate, num_channels, &frame);
+ apm->ProcessStream(&frame);
+ EXPECT_EQ(frame.data()[100], audio_level)
+ << "With factor 1, frame shouldn't be modified.";
+
+ constexpr float gain_factor = 2.f;
+ apm->SetRuntimeSetting(
+ AudioProcessing::RuntimeSetting::CreateCapturePreGain(gain_factor));
+
+ // Process for two frames to have time to ramp up gain.
+ for (int i = 0; i < 2; ++i) {
+ GenerateFixedFrame(audio_level, input_rate, num_channels, &frame);
+ apm->ProcessStream(&frame);
+ }
+ EXPECT_EQ(frame.data()[100], gain_factor * audio_level)
+ << "Frame should be amplified.";
}
} // namespace webrtc