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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 |
11 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h" | 11 #include "webrtc/modules/audio_processing/test/audio_processing_simulator.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <iostream> | 14 #include <iostream> |
15 #include <sstream> | 15 #include <sstream> |
16 #include <string> | 16 #include <string> |
17 #include <utility> | |
17 #include <vector> | 18 #include <vector> |
18 | 19 |
19 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
21 #include "webrtc/base/logging.h" | |
20 #include "webrtc/base/stringutils.h" | 22 #include "webrtc/base/stringutils.h" |
21 #include "webrtc/common_audio/include/audio_util.h" | 23 #include "webrtc/common_audio/include/audio_util.h" |
22 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" | 24 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" |
AleBzk
2017/06/29 11:43:36
Unrelated (merge)
| |
23 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 25 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
26 #include "webrtc/modules/audio_processing/test/fake_recording_device.h" | |
24 | 27 |
25 namespace webrtc { | 28 namespace webrtc { |
26 namespace test { | 29 namespace test { |
27 namespace { | 30 namespace { |
28 | 31 |
32 constexpr FakeRecordingDevice::DeviceKind kDefaultFakeRecDeviceKind = | |
33 FakeRecordingDevice::DeviceKind::IDENTITY; | |
34 | |
29 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) { | 35 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) { |
30 RTC_CHECK_EQ(src.num_channels_, dest->num_channels()); | 36 RTC_CHECK_EQ(src.num_channels_, dest->num_channels()); |
31 RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames()); | 37 RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames()); |
32 // Copy the data from the input buffer. | 38 // Copy the data from the input buffer. |
33 std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_); | 39 std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_); |
34 S16ToFloat(src.data(), tmp.size(), tmp.data()); | 40 S16ToFloat(src.data(), tmp.size(), tmp.data()); |
35 Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_, | 41 Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_, |
36 dest->channels()); | 42 dest->channels()); |
37 } | 43 } |
38 | 44 |
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73 for (size_t ch = 0; ch < dest->num_channels_; ++ch) { | 79 for (size_t ch = 0; ch < dest->num_channels_; ++ch) { |
74 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) { | 80 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) { |
75 dest_data[sample * dest->num_channels_ + ch] = | 81 dest_data[sample * dest->num_channels_ + ch] = |
76 src.channels()[ch][sample] * 32767; | 82 src.channels()[ch][sample] * 32767; |
77 } | 83 } |
78 } | 84 } |
79 } | 85 } |
80 | 86 |
81 AudioProcessingSimulator::AudioProcessingSimulator( | 87 AudioProcessingSimulator::AudioProcessingSimulator( |
82 const SimulationSettings& settings) | 88 const SimulationSettings& settings) |
83 : settings_(settings), worker_queue_("file_writer_task_queue") { | 89 : settings_(settings), |
90 fake_recording_device_(settings.initial_mic_level, | |
91 settings_.simulate_mic_gain | |
92 ? static_cast<FakeRecordingDevice::DeviceKind>( | |
peah-webrtc
2017/06/29 22:04:00
I think you should move the selection of DeviceKin
AleBzk
2017/07/26 13:42:30
Done.
| |
93 *settings.simulated_mic_kind) | |
94 : kDefaultFakeRecDeviceKind), | |
95 worker_queue_("file_writer_task_queue") { | |
AleBzk
2017/06/29 11:43:36
last line is unrelated (merge)
| |
84 if (settings_.ed_graph_output_filename && | 96 if (settings_.ed_graph_output_filename && |
85 settings_.ed_graph_output_filename->size() > 0) { | 97 settings_.ed_graph_output_filename->size() > 0) { |
86 residual_echo_likelihood_graph_writer_.open( | 98 residual_echo_likelihood_graph_writer_.open( |
87 *settings_.ed_graph_output_filename); | 99 *settings_.ed_graph_output_filename); |
88 RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open()); | 100 RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open()); |
89 WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_); | 101 WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_); |
90 } | 102 } |
91 } | 103 } |
92 | 104 |
93 AudioProcessingSimulator::~AudioProcessingSimulator() { | 105 AudioProcessingSimulator::~AudioProcessingSimulator() { |
94 if (residual_echo_likelihood_graph_writer_.is_open()) { | 106 if (residual_echo_likelihood_graph_writer_.is_open()) { |
95 WriteEchoLikelihoodGraphFileFooter(&residual_echo_likelihood_graph_writer_); | 107 WriteEchoLikelihoodGraphFileFooter(&residual_echo_likelihood_graph_writer_); |
96 residual_echo_likelihood_graph_writer_.close(); | 108 residual_echo_likelihood_graph_writer_.close(); |
97 } | 109 } |
98 } | 110 } |
99 | 111 |
100 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() { | 112 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() { |
101 int64_t interval = rtc::TimeNanos() - start_time_; | 113 int64_t interval = rtc::TimeNanos() - start_time_; |
102 proc_time_->sum += interval; | 114 proc_time_->sum += interval; |
103 proc_time_->max = std::max(proc_time_->max, interval); | 115 proc_time_->max = std::max(proc_time_->max, interval); |
104 proc_time_->min = std::min(proc_time_->min, interval); | 116 proc_time_->min = std::min(proc_time_->min, interval); |
105 } | 117 } |
106 | 118 |
107 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { | 119 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) { |
120 if (settings_.aec_dump_input_filename && settings_.simulate_mic_gain) { | |
peah-webrtc
2017/06/29 22:04:00
This is better, but I'd rephrase it as
if (sett
AleBzk
2017/07/26 13:42:30
Done.
| |
121 // When the analog gain is sumulated and an AEC dump is used as input, set | |
122 // the undo level to |aec_dump_mic_level_| to virtually restore the | |
123 // unmodified microphone signal level. | |
124 RTC_DCHECK(aec_dump_mic_level_); | |
125 fake_recording_device_.set_undo_mic_level(aec_dump_mic_level_); | |
126 } | |
127 | |
128 // Optionally use the fake recording device to simulate analog gain. | |
129 if (settings_.simulate_mic_gain) { | |
130 if (fixed_interface) { | |
131 fake_recording_device_.SimulateAnalogGain(&fwd_frame_); | |
132 } else { | |
133 fake_recording_device_.SimulateAnalogGain(in_buf_.get()); | |
134 } | |
135 } | |
136 | |
137 // Notify the current mic level to AGC. | |
138 if (settings_.simulate_mic_gain) { | |
139 RTC_CHECK_EQ(AudioProcessing::kNoError, | |
140 ap_->gain_control()->set_stream_analog_level( | |
peah-webrtc
2017/06/29 22:04:00
No, I don't think this is ok. The fake recording d
AleBzk
2017/07/26 13:42:30
The code below is executed only when the mic gain
| |
141 fake_recording_device_.mic_level())); | |
142 } | |
143 | |
144 // Process the current audio frame. | |
108 if (fixed_interface) { | 145 if (fixed_interface) { |
109 { | 146 { |
110 const auto st = ScopedTimer(mutable_proc_time()); | 147 const auto st = ScopedTimer(mutable_proc_time()); |
111 RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_)); | 148 RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_)); |
112 } | 149 } |
113 CopyFromAudioFrame(fwd_frame_, out_buf_.get()); | 150 CopyFromAudioFrame(fwd_frame_, out_buf_.get()); |
114 } else { | 151 } else { |
115 const auto st = ScopedTimer(mutable_proc_time()); | 152 const auto st = ScopedTimer(mutable_proc_time()); |
116 RTC_CHECK_EQ(AudioProcessing::kNoError, | 153 RTC_CHECK_EQ(AudioProcessing::kNoError, |
117 ap_->ProcessStream(in_buf_->channels(), in_config_, | 154 ap_->ProcessStream(in_buf_->channels(), in_config_, |
118 out_config_, out_buf_->channels())); | 155 out_config_, out_buf_->channels())); |
119 } | 156 } |
120 | 157 |
158 // Store the mic level suggested by AGC. | |
159 fake_recording_device_.set_mic_level( | |
160 ap_->gain_control()->stream_analog_level()); | |
161 | |
121 if (buffer_writer_) { | 162 if (buffer_writer_) { |
122 buffer_writer_->Write(*out_buf_); | 163 buffer_writer_->Write(*out_buf_); |
123 } | 164 } |
124 | 165 |
125 if (residual_echo_likelihood_graph_writer_.is_open()) { | 166 if (residual_echo_likelihood_graph_writer_.is_open()) { |
126 auto stats = ap_->GetStatistics(); | 167 auto stats = ap_->GetStatistics(); |
127 residual_echo_likelihood_graph_writer_ << stats.residual_echo_likelihood | 168 residual_echo_likelihood_graph_writer_ << stats.residual_echo_likelihood |
128 << ", "; | 169 << ", "; |
129 } | 170 } |
130 | 171 |
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188 fwd_frame_.samples_per_channel_ = | 229 fwd_frame_.samples_per_channel_ = |
189 rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond); | 230 rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond); |
190 fwd_frame_.num_channels_ = input_num_channels; | 231 fwd_frame_.num_channels_ = input_num_channels; |
191 | 232 |
192 rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz; | 233 rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz; |
193 rev_frame_.samples_per_channel_ = | 234 rev_frame_.samples_per_channel_ = |
194 rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond); | 235 rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond); |
195 rev_frame_.num_channels_ = reverse_input_num_channels; | 236 rev_frame_.num_channels_ = reverse_input_num_channels; |
196 | 237 |
197 if (settings_.use_verbose_logging) { | 238 if (settings_.use_verbose_logging) { |
239 rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE); | |
240 | |
198 std::cout << "Sample rates:" << std::endl; | 241 std::cout << "Sample rates:" << std::endl; |
199 std::cout << " Forward input: " << input_sample_rate_hz << std::endl; | 242 std::cout << " Forward input: " << input_sample_rate_hz << std::endl; |
200 std::cout << " Forward output: " << output_sample_rate_hz << std::endl; | 243 std::cout << " Forward output: " << output_sample_rate_hz << std::endl; |
201 std::cout << " Reverse input: " << reverse_input_sample_rate_hz | 244 std::cout << " Reverse input: " << reverse_input_sample_rate_hz |
202 << std::endl; | 245 << std::endl; |
203 std::cout << " Reverse output: " << reverse_output_sample_rate_hz | 246 std::cout << " Reverse output: " << reverse_output_sample_rate_hz |
204 << std::endl; | 247 << std::endl; |
205 std::cout << "Number of channels: " << std::endl; | 248 std::cout << "Number of channels: " << std::endl; |
206 std::cout << " Forward input: " << input_num_channels << std::endl; | 249 std::cout << " Forward input: " << input_num_channels << std::endl; |
207 std::cout << " Forward output: " << output_num_channels << std::endl; | 250 std::cout << " Forward output: " << output_num_channels << std::endl; |
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243 static_cast<size_t>(reverse_out_config_.num_channels()))); | 286 static_cast<size_t>(reverse_out_config_.num_channels()))); |
244 reverse_buffer_writer_.reset( | 287 reverse_buffer_writer_.reset( |
245 new ChannelBufferWavWriter(std::move(reverse_out_file))); | 288 new ChannelBufferWavWriter(std::move(reverse_out_file))); |
246 } | 289 } |
247 | 290 |
248 ++output_reset_counter_; | 291 ++output_reset_counter_; |
249 } | 292 } |
250 | 293 |
251 void AudioProcessingSimulator::DestroyAudioProcessor() { | 294 void AudioProcessingSimulator::DestroyAudioProcessor() { |
252 if (settings_.aec_dump_output_filename) { | 295 if (settings_.aec_dump_output_filename) { |
253 ap_->DetachAecDump(); | 296 ap_->DetachAecDump(); |
AleBzk
2017/06/29 11:43:36
Unrelated (merge)
| |
254 } | 297 } |
255 } | 298 } |
256 | 299 |
257 void AudioProcessingSimulator::CreateAudioProcessor() { | 300 void AudioProcessingSimulator::CreateAudioProcessor() { |
258 Config config; | 301 Config config; |
259 AudioProcessing::Config apm_config; | 302 AudioProcessing::Config apm_config; |
260 if (settings_.use_bf && *settings_.use_bf) { | 303 if (settings_.use_bf && *settings_.use_bf) { |
261 config.Set<Beamforming>(new Beamforming( | 304 config.Set<Beamforming>(new Beamforming( |
262 true, ParseArrayGeometry(*settings_.microphone_positions), | 305 true, ParseArrayGeometry(*settings_.microphone_positions), |
263 SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f, | 306 SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f, |
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384 ap_->noise_suppression()->set_level( | 427 ap_->noise_suppression()->set_level( |
385 static_cast<NoiseSuppression::Level>(*settings_.ns_level))); | 428 static_cast<NoiseSuppression::Level>(*settings_.ns_level))); |
386 } | 429 } |
387 | 430 |
388 if (settings_.use_ts) { | 431 if (settings_.use_ts) { |
389 ap_->set_stream_key_pressed(*settings_.use_ts); | 432 ap_->set_stream_key_pressed(*settings_.use_ts); |
390 } | 433 } |
391 | 434 |
392 if (settings_.aec_dump_output_filename) { | 435 if (settings_.aec_dump_output_filename) { |
393 ap_->AttachAecDump(AecDumpFactory::Create( | 436 ap_->AttachAecDump(AecDumpFactory::Create( |
394 *settings_.aec_dump_output_filename, -1, &worker_queue_)); | 437 *settings_.aec_dump_output_filename, -1, &worker_queue_)); |
AleBzk
2017/06/29 11:43:36
Unrelated (merge)
| |
395 } | 438 } |
396 } | 439 } |
397 | 440 |
398 } // namespace test | 441 } // namespace test |
399 } // namespace webrtc | 442 } // namespace webrtc |
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