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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 <algorithm> | |
11 #include <iostream> | 12 #include <iostream> |
13 #include <utility> | |
12 | 14 |
13 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h" | 15 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h" |
14 | 16 |
15 #include "webrtc/base/checks.h" | 17 #include "webrtc/base/checks.h" |
18 #include "webrtc/base/logging.h" | |
19 #include "webrtc/modules/audio_processing/test/fake_recording_device.h" | |
16 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" | 20 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
17 #include "webrtc/test/testsupport/trace_to_stderr.h" | 21 #include "webrtc/test/testsupport/trace_to_stderr.h" |
18 | 22 |
19 namespace webrtc { | 23 namespace webrtc { |
20 namespace test { | 24 namespace test { |
21 namespace { | 25 namespace { |
22 | 26 |
23 // Verify output bitexactness for the fixed interface. | 27 // Verify output bitexactness for the fixed interface. |
24 // TODO(peah): Check whether it would make sense to add a threshold | 28 // TODO(peah): Check whether it would make sense to add a threshold |
25 // to use for checking the bitexactness in a soft manner. | 29 // to use for checking the bitexactness in a soft manner. |
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56 } | 60 } |
57 } | 61 } |
58 } | 62 } |
59 } | 63 } |
60 return true; | 64 return true; |
61 } | 65 } |
62 | 66 |
63 } // namespace | 67 } // namespace |
64 | 68 |
65 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) | 69 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) |
66 : AudioProcessingSimulator(settings) {} | 70 : AudioProcessingSimulator(settings) { |
71 if (settings_.simulate_mic_gain) { | |
72 LOG(LS_VERBOSE) << "Simulating analog mic gain using AEC dump as input " | |
73 << "(the unmodified mic gain level will be virtually restored)"; | |
74 } | |
75 } | |
67 | 76 |
68 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; | 77 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; |
69 | 78 |
70 void AecDumpBasedSimulator::PrepareProcessStreamCall( | 79 void AecDumpBasedSimulator::PrepareProcessStreamCall( |
71 const webrtc::audioproc::Stream& msg, | 80 const webrtc::audioproc::Stream& msg) { |
72 bool* set_stream_analog_level_called) { | |
73 if (msg.has_input_data()) { | 81 if (msg.has_input_data()) { |
74 // Fixed interface processing. | 82 // Fixed interface processing. |
75 // Verify interface invariance. | 83 // Verify interface invariance. |
76 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || | 84 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || |
77 interface_used_ == InterfaceType::kNotSpecified); | 85 interface_used_ == InterfaceType::kNotSpecified); |
78 interface_used_ = InterfaceType::kFixedInterface; | 86 interface_used_ = InterfaceType::kFixedInterface; |
79 | 87 |
80 // Populate input buffer. | 88 // Populate input buffer. |
81 RTC_CHECK_EQ(sizeof(fwd_frame_.data_[0]) * fwd_frame_.samples_per_channel_ * | 89 RTC_CHECK_EQ(sizeof(fwd_frame_.data_[0]) * fwd_frame_.samples_per_channel_ * |
82 fwd_frame_.num_channels_, | 90 fwd_frame_.num_channels_, |
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149 } | 157 } |
150 | 158 |
151 if (!settings_.use_ts) { | 159 if (!settings_.use_ts) { |
152 if (msg.has_keypress()) { | 160 if (msg.has_keypress()) { |
153 ap_->set_stream_key_pressed(msg.keypress()); | 161 ap_->set_stream_key_pressed(msg.keypress()); |
154 } | 162 } |
155 } else { | 163 } else { |
156 ap_->set_stream_key_pressed(*settings_.use_ts); | 164 ap_->set_stream_key_pressed(*settings_.use_ts); |
157 } | 165 } |
158 | 166 |
159 // TODO(peah): Add support for controlling the analog level via the | 167 // Level is always logged in AEC dumps. |
160 // command-line. | 168 RTC_CHECK(msg.has_level()); |
161 if (msg.has_level()) { | 169 |
162 RTC_CHECK_EQ(AudioProcessing::kNoError, | 170 RTC_DCHECK(fake_recording_device_); |
163 ap_->gain_control()->set_stream_analog_level(msg.level())); | 171 if (settings_.simulate_mic_gain) { |
164 *set_stream_analog_level_called = true; | 172 // When the analog gain is simulated, set the undo level to |msg.level()| to |
173 // virtually restore the unmodified microphone signal level. | |
174 fake_recording_device_->set_undo_mic_level(rtc::Optional<int>(msg.level())); | |
165 } else { | 175 } else { |
166 *set_stream_analog_level_called = false; | 176 // When the analog gain is not simulated, the AEC dump level has to be used |
177 // in AudioProcessingSimulator::ProcessStream() - i.e., overriding any value | |
178 // set from a gain controller once the previous audio frame has been | |
179 // analyzed. | |
180 fake_recording_device_->set_mic_level(msg.level()); | |
AleBzk
2017/05/23 13:56:41
@Per: you asked to move this to the parent class a
peah-webrtc
2017/05/23 22:13:20
What I don't like with this approach is that it gi
AleBzk
2017/06/22 10:16:00
Done.
| |
167 } | 181 } |
168 } | 182 } |
169 | 183 |
170 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( | 184 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( |
171 const webrtc::audioproc::Stream& msg) { | 185 const webrtc::audioproc::Stream& msg) { |
172 if (bitexact_output_) { | 186 if (bitexact_output_) { |
173 if (interface_used_ == InterfaceType::kFixedInterface) { | 187 if (interface_used_ == InterfaceType::kFixedInterface) { |
174 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); | 188 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); |
175 } else { | 189 } else { |
176 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); | 190 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); |
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555 } | 569 } |
556 | 570 |
557 SetupBuffersConfigsOutputs( | 571 SetupBuffersConfigsOutputs( |
558 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), | 572 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), |
559 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, | 573 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, |
560 msg.num_reverse_channels(), num_reverse_output_channels); | 574 msg.num_reverse_channels(), num_reverse_output_channels); |
561 } | 575 } |
562 | 576 |
563 void AecDumpBasedSimulator::HandleMessage( | 577 void AecDumpBasedSimulator::HandleMessage( |
564 const webrtc::audioproc::Stream& msg) { | 578 const webrtc::audioproc::Stream& msg) { |
565 bool set_stream_analog_level_called = false; | 579 PrepareProcessStreamCall(msg); |
566 PrepareProcessStreamCall(msg, &set_stream_analog_level_called); | |
567 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); | 580 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); |
568 if (set_stream_analog_level_called) { | |
569 // Call stream analog level to ensure that any side-effects are triggered. | |
570 (void)ap_->gain_control()->stream_analog_level(); | |
571 } | |
572 | |
573 VerifyProcessStreamBitExactness(msg); | 581 VerifyProcessStreamBitExactness(msg); |
574 } | 582 } |
575 | 583 |
576 void AecDumpBasedSimulator::HandleMessage( | 584 void AecDumpBasedSimulator::HandleMessage( |
577 const webrtc::audioproc::ReverseStream& msg) { | 585 const webrtc::audioproc::ReverseStream& msg) { |
578 PrepareReverseProcessStreamCall(msg); | 586 PrepareReverseProcessStreamCall(msg); |
579 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); | 587 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); |
580 } | 588 } |
581 | 589 |
582 } // namespace test | 590 } // namespace test |
583 } // namespace webrtc | 591 } // namespace webrtc |
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