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Side by Side Diff: webrtc/modules/audio_processing/test/audio_processing_simulator.cc

Issue 2486763002: Add support to audioproc_f for running the residual echo detector and producing an echo likelihood … (Closed)
Patch Set: Initial version Created 4 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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54 for (size_t ch = 0; ch < dest->num_channels_; ++ch) { 54 for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
55 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) { 55 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
56 dest->data_[sample * dest->num_channels_ + ch] = 56 dest->data_[sample * dest->num_channels_ + ch] =
57 src.channels()[ch][sample] * 32767; 57 src.channels()[ch][sample] * 32767;
58 } 58 }
59 } 59 }
60 } 60 }
61 61
62 AudioProcessingSimulator::AudioProcessingSimulator( 62 AudioProcessingSimulator::AudioProcessingSimulator(
63 const SimulationSettings& settings) 63 const SimulationSettings& settings)
64 : settings_(settings) {} 64 : settings_(settings) {
65 if (settings_.red_graph_output_filename &&
66 settings_.red_graph_output_filename->size() > 0) {
67 residual_echo_likelihood_graph_.open(*settings_.red_graph_output_filename);
68 residual_echo_likelihood_graph_ << "import numpy as np" << std::endl
peah-webrtc 2016/11/10 10:47:10 Would it be possible to abstract this printout int
ivoc 2016/11/10 15:36:08 Ok, done.
69 << "import matplotlib.pyplot as plt"
70 << std::endl
71 << "y = np.array([";
72 }
73 }
65 74
66 AudioProcessingSimulator::~AudioProcessingSimulator() = default; 75 AudioProcessingSimulator::~AudioProcessingSimulator() {
76 if (residual_echo_likelihood_graph_.is_open()) {
77 residual_echo_likelihood_graph_ << "])" << std::endl
peah-webrtc 2016/11/10 10:47:09 Would it be possible to abstract this printout int
ivoc 2016/11/10 15:36:08 Done.
78 << "x = np.arange(len(y))*.01" << std::endl
79 << "plt.plot(x, y)" << std::endl
80 << "plt.ylabel('Echo likelihood')"
81 << std::endl
82 << "plt.xlabel('Time (s)')" << std::endl
83 << "plt.ylim([0,1])" << std::endl
84 << "plt.show()" << std::endl;
85 residual_echo_likelihood_graph_.close();
86 }
87 }
67 88
68 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() { 89 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
69 int64_t interval = rtc::TimeNanos() - start_time_; 90 int64_t interval = rtc::TimeNanos() - start_time_;
70 proc_time_->sum += interval; 91 proc_time_->sum += interval;
71 proc_time_->max = std::max(proc_time_->max, interval); 92 proc_time_->max = std::max(proc_time_->max, interval);
72 proc_time_->min = std::min(proc_time_->min, interval); 93 proc_time_->min = std::min(proc_time_->min, interval);
73 } 94 }
74 95
75 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { 96 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
76 if (fixed_interface) { 97 if (fixed_interface) {
77 { 98 {
78 const auto st = ScopedTimer(mutable_proc_time()); 99 const auto st = ScopedTimer(mutable_proc_time());
79 RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_)); 100 RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
80 } 101 }
81 CopyFromAudioFrame(fwd_frame_, out_buf_.get()); 102 CopyFromAudioFrame(fwd_frame_, out_buf_.get());
82 } else { 103 } else {
83 const auto st = ScopedTimer(mutable_proc_time()); 104 const auto st = ScopedTimer(mutable_proc_time());
84 RTC_CHECK_EQ(AudioProcessing::kNoError, 105 RTC_CHECK_EQ(AudioProcessing::kNoError,
85 ap_->ProcessStream(in_buf_->channels(), in_config_, 106 ap_->ProcessStream(in_buf_->channels(), in_config_,
86 out_config_, out_buf_->channels())); 107 out_config_, out_buf_->channels()));
87 } 108 }
88 109
89 if (buffer_writer_) { 110 if (buffer_writer_) {
90 buffer_writer_->Write(*out_buf_); 111 buffer_writer_->Write(*out_buf_);
91 } 112 }
92 113
114 if (residual_echo_likelihood_graph_.is_open()) {
115 auto stats = ap_->GetStatistics();
116 residual_echo_likelihood_graph_ << stats.residual_echo_likelihood << ", ";
117 }
118
93 ++num_process_stream_calls_; 119 ++num_process_stream_calls_;
94 } 120 }
95 121
96 void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) { 122 void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
97 if (fixed_interface) { 123 if (fixed_interface) {
98 const auto st = ScopedTimer(mutable_proc_time()); 124 const auto st = ScopedTimer(mutable_proc_time());
99 RTC_CHECK_EQ(AudioProcessing::kNoError, 125 RTC_CHECK_EQ(AudioProcessing::kNoError,
100 ap_->ProcessReverseStream(&rev_frame_)); 126 ap_->ProcessReverseStream(&rev_frame_));
101 CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get()); 127 CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get());
102 128
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238 apm_config.level_controller.enabled = *settings_.use_lc; 264 apm_config.level_controller.enabled = *settings_.use_lc;
239 } 265 }
240 if (settings_.use_refined_adaptive_filter) { 266 if (settings_.use_refined_adaptive_filter) {
241 config.Set<RefinedAdaptiveFilter>( 267 config.Set<RefinedAdaptiveFilter>(
242 new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter)); 268 new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
243 } 269 }
244 config.Set<ExtendedFilter>(new ExtendedFilter( 270 config.Set<ExtendedFilter>(new ExtendedFilter(
245 !settings_.use_extended_filter || *settings_.use_extended_filter)); 271 !settings_.use_extended_filter || *settings_.use_extended_filter));
246 config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic || 272 config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
247 *settings_.use_delay_agnostic)); 273 *settings_.use_delay_agnostic));
274 if (settings_.use_red) {
275 apm_config.residual_echo_detector.enabled = *settings_.use_red;
276 }
248 277
249 ap_.reset(AudioProcessing::Create(config)); 278 ap_.reset(AudioProcessing::Create(config));
250 RTC_CHECK(ap_); 279 RTC_CHECK(ap_);
251 280
252 ap_->ApplyConfig(apm_config); 281 ap_->ApplyConfig(apm_config);
253 282
254 if (settings_.use_aec) { 283 if (settings_.use_aec) {
255 RTC_CHECK_EQ(AudioProcessing::kNoError, 284 RTC_CHECK_EQ(AudioProcessing::kNoError,
256 ap_->echo_cancellation()->Enable(*settings_.use_aec)); 285 ap_->echo_cancellation()->Enable(*settings_.use_aec));
257 } 286 }
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343 size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize; 372 size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
344 RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize); 373 RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
345 RTC_CHECK_EQ(AudioProcessing::kNoError, 374 RTC_CHECK_EQ(AudioProcessing::kNoError,
346 ap_->StartDebugRecording( 375 ap_->StartDebugRecording(
347 settings_.aec_dump_output_filename->c_str(), -1)); 376 settings_.aec_dump_output_filename->c_str(), -1));
348 } 377 }
349 } 378 }
350 379
351 } // namespace test 380 } // namespace test
352 } // namespace webrtc 381 } // namespace webrtc
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