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Activate the pre-amplifier in AudioProcessing.

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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}
This commit is contained in:
Alex Loiko
2018-04-16 16:31:22 +02:00
committed by Commit Bot
parent bfd54ef5cb
commit b5c9a79e68
4 changed files with 79 additions and 7 deletions
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48
modules/audio_processing/audio_processing_impl_unittest.cc
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@ -33,6 +33,24 @@ class MockInitialize : public AudioProcessingImpl {
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, AudioParameterChangeTriggersInit) {
}
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