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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 <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/base/checks.h" | 19 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/stringutils.h" | 20 #include "webrtc/base/stringutils.h" |
21 #include "webrtc/common_audio/include/audio_util.h" | 21 #include "webrtc/common_audio/include/audio_util.h" |
| 22 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h" |
22 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 23 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
23 | 24 |
24 namespace webrtc { | 25 namespace webrtc { |
25 namespace test { | 26 namespace test { |
26 namespace { | 27 namespace { |
27 | 28 |
28 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) { | 29 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) { |
29 RTC_CHECK_EQ(src.num_channels_, dest->num_channels()); | 30 RTC_CHECK_EQ(src.num_channels_, dest->num_channels()); |
30 RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames()); | 31 RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames()); |
31 // Copy the data from the input buffer. | 32 // Copy the data from the input buffer. |
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72 for (size_t ch = 0; ch < dest->num_channels_; ++ch) { | 73 for (size_t ch = 0; ch < dest->num_channels_; ++ch) { |
73 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) { | 74 for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) { |
74 dest_data[sample * dest->num_channels_ + ch] = | 75 dest_data[sample * dest->num_channels_ + ch] = |
75 src.channels()[ch][sample] * 32767; | 76 src.channels()[ch][sample] * 32767; |
76 } | 77 } |
77 } | 78 } |
78 } | 79 } |
79 | 80 |
80 AudioProcessingSimulator::AudioProcessingSimulator( | 81 AudioProcessingSimulator::AudioProcessingSimulator( |
81 const SimulationSettings& settings) | 82 const SimulationSettings& settings) |
82 : settings_(settings) { | 83 : settings_(settings), worker_queue_("file_writer_task_queue") { |
83 if (settings_.ed_graph_output_filename && | 84 if (settings_.ed_graph_output_filename && |
84 settings_.ed_graph_output_filename->size() > 0) { | 85 settings_.ed_graph_output_filename->size() > 0) { |
85 residual_echo_likelihood_graph_writer_.open( | 86 residual_echo_likelihood_graph_writer_.open( |
86 *settings_.ed_graph_output_filename); | 87 *settings_.ed_graph_output_filename); |
87 RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open()); | 88 RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open()); |
88 WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_); | 89 WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_); |
89 } | 90 } |
90 } | 91 } |
91 | 92 |
92 AudioProcessingSimulator::~AudioProcessingSimulator() { | 93 AudioProcessingSimulator::~AudioProcessingSimulator() { |
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242 static_cast<size_t>(reverse_out_config_.num_channels()))); | 243 static_cast<size_t>(reverse_out_config_.num_channels()))); |
243 reverse_buffer_writer_.reset( | 244 reverse_buffer_writer_.reset( |
244 new ChannelBufferWavWriter(std::move(reverse_out_file))); | 245 new ChannelBufferWavWriter(std::move(reverse_out_file))); |
245 } | 246 } |
246 | 247 |
247 ++output_reset_counter_; | 248 ++output_reset_counter_; |
248 } | 249 } |
249 | 250 |
250 void AudioProcessingSimulator::DestroyAudioProcessor() { | 251 void AudioProcessingSimulator::DestroyAudioProcessor() { |
251 if (settings_.aec_dump_output_filename) { | 252 if (settings_.aec_dump_output_filename) { |
252 RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording()); | 253 ap_->DetachAecDump(); |
253 } | 254 } |
254 } | 255 } |
255 | 256 |
256 void AudioProcessingSimulator::CreateAudioProcessor() { | 257 void AudioProcessingSimulator::CreateAudioProcessor() { |
257 Config config; | 258 Config config; |
258 AudioProcessing::Config apm_config; | 259 AudioProcessing::Config apm_config; |
259 if (settings_.use_bf && *settings_.use_bf) { | 260 if (settings_.use_bf && *settings_.use_bf) { |
260 config.Set<Beamforming>(new Beamforming( | 261 config.Set<Beamforming>(new Beamforming( |
261 true, ParseArrayGeometry(*settings_.microphone_positions), | 262 true, ParseArrayGeometry(*settings_.microphone_positions), |
262 SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f, | 263 SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f, |
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382 AudioProcessing::kNoError, | 383 AudioProcessing::kNoError, |
383 ap_->noise_suppression()->set_level( | 384 ap_->noise_suppression()->set_level( |
384 static_cast<NoiseSuppression::Level>(*settings_.ns_level))); | 385 static_cast<NoiseSuppression::Level>(*settings_.ns_level))); |
385 } | 386 } |
386 | 387 |
387 if (settings_.use_ts) { | 388 if (settings_.use_ts) { |
388 ap_->set_stream_key_pressed(*settings_.use_ts); | 389 ap_->set_stream_key_pressed(*settings_.use_ts); |
389 } | 390 } |
390 | 391 |
391 if (settings_.aec_dump_output_filename) { | 392 if (settings_.aec_dump_output_filename) { |
392 size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize; | 393 ap_->AttachAecDump(AecDumpFactory::Create( |
393 RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize); | 394 *settings_.aec_dump_output_filename, -1, &worker_queue_)); |
394 RTC_CHECK_EQ(AudioProcessing::kNoError, | |
395 ap_->StartDebugRecording( | |
396 settings_.aec_dump_output_filename->c_str(), -1)); | |
397 } | 395 } |
398 } | 396 } |
399 | 397 |
400 } // namespace test | 398 } // namespace test |
401 } // namespace webrtc | 399 } // namespace webrtc |
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