Add AudioProcessingCaptureStats and a level estimator replacement
This adds an interface for accessing stats on the capture stream, and adds a level estimator to report one of the stats. Bug: webrtc:9947 Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb Reviewed-on: https://webrtc-review.googlesource.com/c/109587 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25786}
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@ -2801,4 +2801,42 @@ TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
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EXPECT_FALSE(stats.delay_median_ms);
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EXPECT_FALSE(stats.delay_standard_deviation_ms);
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}
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TEST(ApmStatistics, ReportOutputRmsDbfs) {
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ProcessingConfig processing_config = {
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{{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
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AudioProcessing::Config config;
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// Set up an audioframe.
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AudioFrame frame;
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frame.num_channels_ = 1;
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SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate48kHz);
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// Fill the audio frame with a sawtooth pattern.
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int16_t* ptr = frame.mutable_data();
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for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
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ptr[i] = 10000 * ((i % 3) - 1);
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}
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std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
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apm->Initialize(processing_config);
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// If not enabled, no metric should be reported.
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EXPECT_EQ(apm->ProcessStream(&frame), 0);
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EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
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// If enabled, metrics should be reported.
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config.level_estimation.enabled = true;
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apm->ApplyConfig(config);
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EXPECT_EQ(apm->ProcessStream(&frame), 0);
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auto stats = apm->GetStatistics(false);
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EXPECT_TRUE(stats.output_rms_dbfs);
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EXPECT_GE(*stats.output_rms_dbfs, 0);
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// If re-disabled, the value is again not reported.
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config.level_estimation.enabled = false;
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apm->ApplyConfig(config);
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EXPECT_EQ(apm->ProcessStream(&frame), 0);
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EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
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}
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} // namespace webrtc
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