Chromium Code Reviews| OLD | NEW | 
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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 #include "webrtc/modules/audio_processing/aec3/echo_canceller3.h" | 10 #include "webrtc/modules/audio_processing/aec3/echo_canceller3.h" | 
| 11 | 11 | 
| 12 #include "webrtc/base/atomicops.h" | 12 #include "webrtc/base/atomicops.h" | 
| 13 #include "webrtc/system_wrappers/include/logging.h" | 13 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | 
| 14 | 14 | 
| 15 namespace webrtc { | 15 namespace webrtc { | 
| 16 | 16 | 
| 17 namespace { | |
| 18 | |
| 19 bool DetectSaturation(rtc::ArrayView<const float> y) { | |
| 20 for (auto y_k : y) { | |
| 21 if (y_k >= 32767.0f || y_k <= -32768.0f) { | |
| 22 return true; | |
| 23 } | |
| 24 } | |
| 25 return false; | |
| 26 } | |
| 27 | |
| 28 void FillSubFrameView(AudioBuffer* frame, | |
| 29 size_t sub_frame_index, | |
| 30 rtc::ArrayView<rtc::ArrayView<float>> sub_frame_view) { | |
| 
 
aleloi
2016/12/20 15:55:35
Can this FillSubFrameView be implemented in terms
 
peah-webrtc
2016/12/21 23:13:51
I changed this to instead used a vector stored on
 
 | |
| 31 for (size_t k = 0; k < frame->num_bands(); ++k) { | |
| 32 sub_frame_view[k] = rtc::ArrayView<float>( | |
| 33 &frame->split_bands_f(0)[k][sub_frame_index * kSubFrameLength], | |
| 34 kSubFrameLength); | |
| 35 } | |
| 36 } | |
| 37 | |
| 38 void FillSubFrameView( | |
| 39 AudioBuffer* frame, | |
| 40 size_t sub_frame_index, | |
| 41 rtc::ArrayView<rtc::ArrayView<const float>> sub_frame_view) { | |
| 42 for (size_t k = 0; k < frame->num_bands(); ++k) { | |
| 43 sub_frame_view[k] = rtc::ArrayView<const float>( | |
| 44 &frame->split_bands_f(0)[k][sub_frame_index * kSubFrameLength], | |
| 45 kSubFrameLength); | |
| 46 } | |
| 47 } | |
| 48 | |
| 49 void FillSubFrameView( | |
| 50 const std::vector<std::vector<float>>& frame, | |
| 51 size_t sub_frame_index, | |
| 52 rtc::ArrayView<rtc::ArrayView<const float>> sub_frame_view) { | |
| 53 for (size_t k = 0; k < frame.size(); ++k) { | |
| 54 sub_frame_view[k] = rtc::ArrayView<const float>( | |
| 55 &frame[k][sub_frame_index * kSubFrameLength], kSubFrameLength); | |
| 56 } | |
| 57 } | |
| 58 | |
| 59 void ProcessCaptureFrameContent(AudioBuffer* capture, | |
| 60 bool known_echo_path_change, | |
| 61 bool saturated_microphone_signal, | |
| 62 size_t sub_frame_index, | |
| 63 FrameBlocker* capture_blocker, | |
| 64 BlockFramer* output_framer, | |
| 65 BlockProcessor* block_processor, | |
| 66 std::vector<std::vector<float>>* block) { | |
| 67 // Array of ArrayViews. | |
| 68 rtc::ArrayView<float> sub_frame_view_data[block->size()]; | |
| 
 
hlundin-webrtc
2016/12/20 15:10:35
This is not pretty. Sorry...
Is there any way we c
 
aleloi
2016/12/21 10:07:45
Parts of the complexity is here because ArrayViews
 
peah-webrtc
2016/12/21 23:13:50
I changed to using a vector that is cached on the
 
peah-webrtc
2016/12/21 23:13:51
Agree. I instead solved this by using a vector sto
 
 | |
| 69 rtc::ArrayView<const float> sub_frame_const_view_data[block->size()]; | |
| 70 auto sub_frame_view = rtc::ArrayView<rtc::ArrayView<float>>( | |
| 71 &sub_frame_view_data[0], block->size()); | |
| 72 auto sub_frame_const_view = rtc::ArrayView<rtc::ArrayView<const float>>( | |
| 73 &sub_frame_const_view_data[0], block->size()); | |
| 74 | |
| 75 FillSubFrameView(capture, sub_frame_index, sub_frame_view); | |
| 
 
aleloi
2016/12/21 10:07:45
Better to move creation and filling of sub_frame_v
 
peah-webrtc
2016/12/21 23:13:51
Agree.
With the new change, it looks better.
PTAL
 
 | |
| 76 FillSubFrameView(capture, sub_frame_index, sub_frame_const_view); | |
| 77 | |
| 78 capture_blocker->InsertSubFrameAndExtractBlock(sub_frame_const_view, block); | |
| 
 
aleloi
2016/12/21 10:07:45
Is there a type error if a non-const view is passe
 
peah-webrtc
2016/12/21 23:13:50
Yes, there was a type error. I don't think the aut
 
 | |
| 79 block_processor->ProcessCapture(known_echo_path_change, | |
| 80 saturated_microphone_signal, block); | |
| 81 output_framer->InsertBlockAndExtractSubFrame(*block, sub_frame_view); | |
| 82 } | |
| 83 | |
| 84 void ProcessRemainingCaptureFrameContent( | |
| 85 bool known_echo_path_change, | |
| 86 bool saturated_microphone_signal, | |
| 87 FrameBlocker* capture_blocker, | |
| 88 BlockFramer* output_framer, | |
| 89 BlockProcessor* block_processor, | |
| 90 std::vector<std::vector<float>>* block) { | |
| 91 if (!capture_blocker->IsBlockAvailable()) { | |
| 92 return; | |
| 93 } | |
| 94 | |
| 95 capture_blocker->ExtractBlock(block); | |
| 96 block_processor->ProcessCapture(known_echo_path_change, | |
| 97 saturated_microphone_signal, block); | |
| 98 output_framer->InsertBlock(*block); | |
| 99 } | |
| 100 | |
| 101 bool BufferRenderFrameContent( | |
| 102 const std::vector<std::vector<float>>& render_frame, | |
| 103 size_t sub_frame_index, | |
| 104 FrameBlocker* render_blocker, | |
| 105 BlockProcessor* block_processor, | |
| 106 std::vector<std::vector<float>>* block) { | |
| 107 // Array of ArrayViews. | |
| 108 rtc::ArrayView<const float> sub_frame_const_view_data[block->size()]; | |
| 109 auto sub_frame_const_view = rtc::ArrayView<rtc::ArrayView<const float>>( | |
| 110 &sub_frame_const_view_data[0], block->size()); | |
| 111 | |
| 112 FillSubFrameView(render_frame, sub_frame_index, sub_frame_const_view); | |
| 113 | |
| 114 render_blocker->InsertSubFrameAndExtractBlock(sub_frame_const_view, block); | |
| 115 return block_processor->BufferRender(block); | |
| 116 } | |
| 117 | |
| 118 bool BufferRemainingRenderFrameContent(FrameBlocker* render_blocker, | |
| 119 BlockProcessor* block_processor, | |
| 120 std::vector<std::vector<float>>* block) { | |
| 121 if (!render_blocker->IsBlockAvailable()) { | |
| 122 return false; | |
| 123 } | |
| 124 render_blocker->ExtractBlock(block); | |
| 125 return block_processor->BufferRender(block); | |
| 126 } | |
| 127 | |
| 128 void CopyAudioBufferIntoFrame(AudioBuffer* buffer, | |
| 129 size_t num_bands, | |
| 130 size_t frame_length, | |
| 131 std::vector<std::vector<float>>* frame) { | |
| 132 RTC_DCHECK_EQ(num_bands, frame->size()); | |
| 133 for (size_t i = 0; i < num_bands; ++i) { | |
| 134 rtc::ArrayView<float> buffer_view(&buffer->split_bands_f(0)[i][0], | |
| 135 frame_length); | |
| 136 std::copy(buffer_view.begin(), buffer_view.end(), (*frame)[i].begin()); | |
| 137 } | |
| 138 } | |
| 139 | |
| 140 // [B,A] = butter(2,100/4000,'high') | |
| 141 const CascadedBiQuadFilter::BiQuadCoefficients | |
| 142 kHighPassFilterCoefficients_8kHz = { | |
| 143 {0.945976856002790, -1.891953712005580, 0.945976856002790}, | |
| 144 {-1.889033079394525, 0.894874344616636}}; | |
| 145 const int kNumberOfHighPassBiQuads_8kHz = 1; | |
| 146 | |
| 147 // [B,A] = butter(2,100/8000,'high') | |
| 148 const CascadedBiQuadFilter::BiQuadCoefficients | |
| 149 kHighPassFilterCoefficients_16kHz = { | |
| 150 {0.972613898499844, -1.945227796999688, 0.972613898499844}, | |
| 151 {-1.944477657767094, 0.945977936232282}}; | |
| 152 const int kNumberOfHighPassBiQuads_16kHz = 1; | |
| 153 | |
| 154 static const size_t kRenderTransferQueueSize = 30; | |
| 155 | |
| 156 } // namespace | |
| 157 | |
| 158 class EchoCanceller3::RenderWriter { | |
| 159 public: | |
| 160 RenderWriter(ApmDataDumper* data_dumper, | |
| 161 SwapQueue<std::vector<std::vector<float>>, | |
| 162 Aec3RenderQueueItemVerifier>* render_transfer_queue, | |
| 163 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter, | |
| 164 int sample_rate_hz, | |
| 165 int frame_length, | |
| 166 int num_bands); | |
| 167 ~RenderWriter(); | |
| 168 bool Insert(AudioBuffer* render); | |
| 169 | |
| 170 private: | |
| 171 ApmDataDumper* data_dumper_; | |
| 172 const int sample_rate_hz_; | |
| 173 const size_t frame_length_; | |
| 174 const int num_bands_; | |
| 175 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter_; | |
| 176 std::vector<std::vector<float>> render_queue_input_frame_; | |
| 177 SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier>* | |
| 178 render_transfer_queue_; | |
| 179 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderWriter); | |
| 180 }; | |
| 181 | |
| 182 EchoCanceller3::RenderWriter::RenderWriter( | |
| 183 ApmDataDumper* data_dumper, | |
| 184 SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier>* | |
| 185 render_transfer_queue, | |
| 186 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter, | |
| 187 int sample_rate_hz, | |
| 188 int frame_length, | |
| 189 int num_bands) | |
| 190 : data_dumper_(data_dumper), | |
| 191 sample_rate_hz_(sample_rate_hz), | |
| 192 frame_length_(frame_length), | |
| 193 num_bands_(num_bands), | |
| 194 render_highpass_filter_(std::move(render_highpass_filter)), | |
| 195 render_queue_input_frame_(num_bands_, | |
| 196 std::vector<float>(frame_length_, 0.f)), | |
| 197 render_transfer_queue_(render_transfer_queue) { | |
| 198 RTC_DCHECK(data_dumper); | |
| 199 } | |
| 200 | |
| 201 EchoCanceller3::RenderWriter::~RenderWriter() = default; | |
| 202 | |
| 203 bool EchoCanceller3::RenderWriter::Insert(AudioBuffer* input) { | |
| 204 RTC_DCHECK_EQ(1, input->num_channels()); | |
| 205 RTC_DCHECK_EQ(frame_length_, input->num_frames_per_band()); | |
| 206 data_dumper_->DumpWav("aec3_render_input", frame_length_, | |
| 207 &input->split_bands_f(0)[0][0], | |
| 208 LowestBandRate(sample_rate_hz_), 1); | |
| 209 | |
| 210 CopyAudioBufferIntoFrame(input, num_bands_, frame_length_, | |
| 211 &render_queue_input_frame_); | |
| 212 | |
| 213 if (render_highpass_filter_) { | |
| 214 render_highpass_filter_->Process(render_queue_input_frame_[0]); | |
| 215 } | |
| 216 | |
| 217 return render_transfer_queue_->Insert(&render_queue_input_frame_); | |
| 218 } | |
| 219 | |
| 17 int EchoCanceller3::instance_count_ = 0; | 220 int EchoCanceller3::instance_count_ = 0; | 
| 18 | 221 | 
| 19 EchoCanceller3::EchoCanceller3(int sample_rate_hz, bool use_anti_hum_filter) { | 222 EchoCanceller3::EchoCanceller3(int sample_rate_hz, bool use_highpass_filter) | 
| 20 int band_sample_rate_hz = (sample_rate_hz == 8000 ? sample_rate_hz : 16000); | 223 : EchoCanceller3(sample_rate_hz, | 
| 21 frame_length_ = rtc::CheckedDivExact(band_sample_rate_hz, 100); | 224 use_highpass_filter, | 
| 22 | 225 BlockProcessor::Create(sample_rate_hz, | 
| 23 LOG(LS_INFO) << "AEC3 created : " | 226 NumBandsForRate(sample_rate_hz))) {} | 
| 24 << "{ instance_count: " << instance_count_ << "}"; | 227 EchoCanceller3::EchoCanceller3(int sample_rate_hz, | 
| 228 bool use_highpass_filter, | |
| 229 BlockProcessor* block_processor) | |
| 230 : data_dumper_(new ApmDataDumper(instance_count_)), | |
| 231 sample_rate_hz_(sample_rate_hz), | |
| 232 num_bands_(NumBandsForRate(sample_rate_hz_)), | |
| 233 frame_length_(rtc::CheckedDivExact(LowestBandRate(sample_rate_hz_), 100)), | |
| 234 output_framer_(num_bands_), | |
| 235 capture_blocker_(num_bands_), | |
| 236 render_blocker_(num_bands_), | |
| 237 render_transfer_queue_( | |
| 238 kRenderTransferQueueSize, | |
| 239 std::vector<std::vector<float>>( | |
| 240 num_bands_, | |
| 241 std::vector<float>(frame_length_, 0.f)), | |
| 242 Aec3RenderQueueItemVerifier(num_bands_, frame_length_)), | |
| 243 block_processor_(block_processor), | |
| 244 render_queue_output_frame_(num_bands_, | |
| 245 std::vector<float>(frame_length_, 0.f)), | |
| 246 block_(num_bands_, std::vector<float>(kBlockSize, 0.f)) { | |
| 247 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter; | |
| 248 if (use_highpass_filter) { | |
| 249 render_highpass_filter.reset(new CascadedBiQuadFilter( | |
| 250 sample_rate_hz_ == 8000 ? kHighPassFilterCoefficients_8kHz | |
| 251 : kHighPassFilterCoefficients_16kHz, | |
| 252 sample_rate_hz_ == 8000 ? kNumberOfHighPassBiQuads_8kHz | |
| 253 : kNumberOfHighPassBiQuads_16kHz)); | |
| 254 capture_highpass_filter_.reset(new CascadedBiQuadFilter( | |
| 255 sample_rate_hz_ == 8000 ? kHighPassFilterCoefficients_8kHz | |
| 256 : kHighPassFilterCoefficients_16kHz, | |
| 257 sample_rate_hz_ == 8000 ? kNumberOfHighPassBiQuads_8kHz | |
| 258 : kNumberOfHighPassBiQuads_16kHz)); | |
| 259 } else { | |
| 260 render_highpass_filter.reset(nullptr); | |
| 
 
ivoc
2016/12/21 13:04:05
Is this really necessary? Isn't it set to nullptr
 
peah-webrtc
2016/12/21 23:13:51
I'm not sure. The issue was that there was an expl
 
ivoc
2016/12/22 13:38:13
I'm pretty sure that it's not needed, the default
 
peah-webrtc
2017/01/02 08:45:10
Thanks! I agree.
Done.
 
 | |
| 261 capture_highpass_filter_.reset(nullptr); | |
| 262 } | |
| 263 | |
| 264 render_writer_.reset( | |
| 265 new RenderWriter(data_dumper_.get(), &render_transfer_queue_, | |
| 266 std::move(render_highpass_filter), sample_rate_hz_, | |
| 267 frame_length_, num_bands_)); | |
| 268 | |
| 269 RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000); | |
| 270 RTC_DCHECK_GE(kMaxNumBands, num_bands_); | |
| 25 instance_count_ = rtc::AtomicOps::Increment(&instance_count_); | 271 instance_count_ = rtc::AtomicOps::Increment(&instance_count_); | 
| 26 } | 272 } | 
| 27 | 273 | 
| 28 EchoCanceller3::~EchoCanceller3() = default; | 274 EchoCanceller3::~EchoCanceller3() = default; | 
| 29 | 275 | 
| 30 bool EchoCanceller3::AnalyzeRender(AudioBuffer* render) { | 276 bool EchoCanceller3::AnalyzeRender(AudioBuffer* render) { | 
| 31 RTC_DCHECK_EQ(1u, render->num_channels()); | 277 return render_writer_->Insert(render); | 
| 32 RTC_DCHECK_EQ(frame_length_, render->num_frames_per_band()); | 278 } | 
| 33 return true; | 279 | 
| 34 } | 280 void EchoCanceller3::AnalyzeCapture(AudioBuffer* capture) { | 
| 35 | 281 data_dumper_->DumpWav("aec3_capture_analyze_input", frame_length_, | 
| 36 void EchoCanceller3::AnalyzeCapture(AudioBuffer* capture) {} | 282 capture->channels_f()[0], sample_rate_hz_, 1); | 
| 283 | |
| 284 saturated_microphone_signal_ = false; | |
| 285 for (size_t k = 0; k < capture->num_channels(); ++k) { | |
| 286 saturated_microphone_signal_ |= | |
| 287 DetectSaturation(rtc::ArrayView<const float>(capture->channels_f()[k], | |
| 288 capture->num_frames())); | |
| 289 if (saturated_microphone_signal_) { | |
| 290 break; | |
| 291 } | |
| 292 } | |
| 293 } | |
| 37 | 294 | 
| 38 void EchoCanceller3::ProcessCapture(AudioBuffer* capture, | 295 void EchoCanceller3::ProcessCapture(AudioBuffer* capture, | 
| 39 bool known_echo_path_change) { | 296 bool known_echo_path_change) { | 
| 40 RTC_DCHECK_EQ(1u, capture->num_channels()); | 297 RTC_DCHECK_EQ(1u, capture->num_channels()); | 
| 41 RTC_DCHECK_EQ(frame_length_, capture->num_frames_per_band()); | 298 RTC_DCHECK_EQ(frame_length_, capture->num_frames_per_band()); | 
| 299 | |
| 300 rtc::ArrayView<float> capture_lower_band = | |
| 301 rtc::ArrayView<float>(&capture->split_bands_f(0)[0][0], frame_length_); | |
| 302 | |
| 303 data_dumper_->DumpWav("aec3_capture_input", capture_lower_band, | |
| 304 LowestBandRate(sample_rate_hz_), 1); | |
| 305 | |
| 306 bool render_buffer_overrun = EmptyRenderQueue(); | |
| 307 RTC_DCHECK(!render_buffer_overrun); | |
| 308 | |
| 309 if (capture_highpass_filter_) { | |
| 310 capture_highpass_filter_->Process(capture_lower_band); | |
| 311 } | |
| 312 | |
| 313 ProcessCaptureFrameContent(capture, known_echo_path_change, | |
| 314 saturated_microphone_signal_, 0, &capture_blocker_, | |
| 315 &output_framer_, block_processor_.get(), &block_); | |
| 316 | |
| 317 if (sample_rate_hz_ != 8000) { | |
| 318 ProcessCaptureFrameContent( | |
| 319 capture, known_echo_path_change, saturated_microphone_signal_, 1, | |
| 320 &capture_blocker_, &output_framer_, block_processor_.get(), &block_); | |
| 321 } | |
| 322 | |
| 323 ProcessRemainingCaptureFrameContent( | |
| 324 known_echo_path_change, saturated_microphone_signal_, &capture_blocker_, | |
| 325 &output_framer_, block_processor_.get(), &block_); | |
| 326 | |
| 327 data_dumper_->DumpWav("aec3_capture_output", frame_length_, | |
| 328 &capture->split_bands_f(0)[0][0], | |
| 329 LowestBandRate(sample_rate_hz_), 1); | |
| 42 } | 330 } | 
| 43 | 331 | 
| 44 std::string EchoCanceller3::ToString( | 332 std::string EchoCanceller3::ToString( | 
| 45 const AudioProcessing::Config::EchoCanceller3& config) { | 333 const AudioProcessing::Config::EchoCanceller3& config) { | 
| 46 std::stringstream ss; | 334 std::stringstream ss; | 
| 47 ss << "{" | 335 ss << "{" | 
| 48 << "enabled: " << (config.enabled ? "true" : "false") << "}"; | 336 << "enabled: " << (config.enabled ? "true" : "false") << "}"; | 
| 49 return ss.str(); | 337 return ss.str(); | 
| 50 } | 338 } | 
| 51 | 339 | 
| 52 bool EchoCanceller3::Validate( | 340 bool EchoCanceller3::Validate( | 
| 53 const AudioProcessing::Config::EchoCanceller3& config) { | 341 const AudioProcessing::Config::EchoCanceller3& config) { | 
| 54 return true; | 342 return true; | 
| 55 } | 343 } | 
| 56 | 344 | 
| 345 bool EchoCanceller3::EmptyRenderQueue() { | |
| 346 bool render_buffer_overrun = false; | |
| 347 bool frame_to_buffer = | |
| 348 render_transfer_queue_.Remove(&render_queue_output_frame_); | |
| 349 while (frame_to_buffer) { | |
| 350 render_buffer_overrun |= BufferRenderFrameContent( | |
| 351 render_queue_output_frame_, 0, &render_blocker_, block_processor_.get(), | |
| 352 &block_); | |
| 353 | |
| 354 if (sample_rate_hz_ != 8000) { | |
| 355 render_buffer_overrun |= BufferRenderFrameContent( | |
| 356 render_queue_output_frame_, 1, &render_blocker_, | |
| 357 block_processor_.get(), &block_); | |
| 358 } | |
| 359 | |
| 360 render_buffer_overrun |= BufferRemainingRenderFrameContent( | |
| 361 &render_blocker_, block_processor_.get(), &block_); | |
| 362 | |
| 363 frame_to_buffer = | |
| 364 render_transfer_queue_.Remove(&render_queue_output_frame_); | |
| 365 } | |
| 366 return render_buffer_overrun; | |
| 367 } | |
| 368 | |
| 57 } // namespace webrtc | 369 } // namespace webrtc | 
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