f3aa6326b8
This CL replaces the use of the ExperimentalAgc config with using the new config format. Beyond that, some further changes were made to how the analog and digital AGCs are initialized/called. While these can be made in a separate CL, I believe the code changes becomes more clear by bundling those with the replacement of the ExperimentalAgc config. TBR: saza@webrtc.org Bug: webrtc:5298 Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30149}
394 lines
16 KiB
C++
394 lines
16 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/gain_control_impl.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "modules/audio_processing/test/bitexactness_tools.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 100;
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void ProcessOneFrame(int sample_rate_hz,
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AudioBuffer* render_audio_buffer,
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AudioBuffer* capture_audio_buffer,
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GainControlImpl* gain_controller) {
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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render_audio_buffer->SplitIntoFrequencyBands();
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capture_audio_buffer->SplitIntoFrequencyBands();
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}
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std::vector<int16_t> render_audio;
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GainControlImpl::PackRenderAudioBuffer(*render_audio_buffer, &render_audio);
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gain_controller->ProcessRenderAudio(render_audio);
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gain_controller->AnalyzeCaptureAudio(*capture_audio_buffer);
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gain_controller->ProcessCaptureAudio(capture_audio_buffer, false);
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if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
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capture_audio_buffer->MergeFrequencyBands();
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}
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}
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void SetupComponent(int sample_rate_hz,
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GainControl::Mode mode,
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int target_level_dbfs,
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int stream_analog_level,
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int compression_gain_db,
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bool enable_limiter,
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int analog_level_min,
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int analog_level_max,
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GainControlImpl* gain_controller) {
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gain_controller->Initialize(1, sample_rate_hz);
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GainControl* gc = static_cast<GainControl*>(gain_controller);
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gc->set_mode(mode);
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gc->set_stream_analog_level(stream_analog_level);
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gc->set_target_level_dbfs(target_level_dbfs);
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gc->set_compression_gain_db(compression_gain_db);
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gc->enable_limiter(enable_limiter);
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gc->set_analog_level_limits(analog_level_min, analog_level_max);
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}
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void RunBitExactnessTest(int sample_rate_hz,
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size_t num_channels,
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GainControl::Mode mode,
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int target_level_dbfs,
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int stream_analog_level,
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int compression_gain_db,
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bool enable_limiter,
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int analog_level_min,
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int analog_level_max,
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int achieved_stream_analog_level_reference,
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rtc::ArrayView<const float> output_reference) {
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GainControlImpl gain_controller;
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SetupComponent(sample_rate_hz, mode, target_level_dbfs, stream_analog_level,
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compression_gain_db, enable_limiter, analog_level_min,
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analog_level_max, &gain_controller);
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const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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const StreamConfig render_config(sample_rate_hz, num_channels, false);
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AudioBuffer render_buffer(
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render_config.sample_rate_hz(), render_config.num_channels(),
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render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
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test::InputAudioFile render_file(
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test::GetApmRenderTestVectorFileName(sample_rate_hz));
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std::vector<float> render_input(samples_per_channel * num_channels);
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const StreamConfig capture_config(sample_rate_hz, num_channels, false);
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AudioBuffer capture_buffer(
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capture_config.sample_rate_hz(), capture_config.num_channels(),
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capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
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test::InputAudioFile capture_file(
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test::GetApmCaptureTestVectorFileName(sample_rate_hz));
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std::vector<float> capture_input(samples_per_channel * num_channels);
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for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&render_file, render_input);
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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ProcessOneFrame(sample_rate_hz, &render_buffer, &capture_buffer,
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&gain_controller);
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}
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// Extract and verify the test results.
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std::vector<float> capture_output;
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test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
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&capture_output);
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EXPECT_EQ(achieved_stream_analog_level_reference,
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gain_controller.stream_analog_level());
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// Compare the output with the reference. Only the first values of the output
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// from last frame processed are compared in order not having to specify all
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// preceeding frames as testvectors. As the algorithm being tested has a
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// memory, testing only the last frame implicitly also tests the preceeding
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// frames.
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const float kElementErrorBound = 1.0f / 32768.0f;
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EXPECT_TRUE(test::VerifyDeinterleavedArray(
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capture_config.num_frames(), capture_config.num_channels(),
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output_reference, capture_output, kElementErrorBound));
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}
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} // namespace
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// TODO(peah): Activate all these tests for ARM and ARM64 once the issue on the
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// Chromium ARM and ARM64 boths have been identified. This is tracked in the
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// issue https://bugs.chromium.org/p/webrtc/issues/detail?id=5711.
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.027313f, -0.015900f, -0.028107f,
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-0.027313f, -0.015900f, -0.028107f};
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RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.003967f, -0.002777f, -0.001770f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.015411f, -0.008972f, -0.015839f,
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-0.015411f, -0.008972f, -0.015839f};
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RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.006134f, -0.005524f, -0.005005f};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.006134f, -0.005524f, -0.005005};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.011749f, -0.008270f, -0.005219f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.048896f, -0.028479f, -0.050345f,
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-0.048896f, -0.028479f, -0.050345f};
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RunBitExactnessTest(16000, 2, GainControl::Mode::kFixedDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.018158f, -0.016357f, -0.014832f};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 50;
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const float kOutputReference[] = {-0.018158f, -0.016357f, -0.014832f};
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RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 12;
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const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 10, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
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#endif
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const int kStreamAnalogLevelReference = 100;
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const float kOutputReference[] = {-0.003998f, -0.002808f, -0.001770f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 100, 5,
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true, 70, 80, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 100;
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const float kOutputReference[] = {-0.004028f, -0.002838f, -0.001770f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100, 5,
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false, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 100;
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const float kOutputReference[] = {-0.008728f, -0.006134f, -0.003845f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 40, 100, 5,
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true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
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defined(WEBRTC_ANDROID))
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TEST(GainControlBitExactnessTest,
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Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
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#else
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TEST(GainControlBitExactnessTest,
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DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
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#endif
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const int kStreamAnalogLevelReference = 100;
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const float kOutputReference[] = {-0.005859f, -0.004120f, -0.002594f};
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RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100,
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30, true, 0, 100, kStreamAnalogLevelReference,
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kOutputReference);
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}
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} // namespace webrtc
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