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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" |
16 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" | 18 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
17 #include "webrtc/test/testsupport/trace_to_stderr.h" | 19 #include "webrtc/test/testsupport/trace_to_stderr.h" |
18 | 20 |
19 namespace webrtc { | 21 namespace webrtc { |
20 namespace test { | 22 namespace test { |
21 namespace { | 23 namespace { |
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61 } | 63 } |
62 | 64 |
63 } // namespace | 65 } // namespace |
64 | 66 |
65 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) | 67 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) |
66 : AudioProcessingSimulator(settings) {} | 68 : AudioProcessingSimulator(settings) {} |
67 | 69 |
68 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; | 70 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; |
69 | 71 |
70 void AecDumpBasedSimulator::PrepareProcessStreamCall( | 72 void AecDumpBasedSimulator::PrepareProcessStreamCall( |
71 const webrtc::audioproc::Stream& msg, | 73 const webrtc::audioproc::Stream& msg) { |
72 bool* set_stream_analog_level_called) { | |
73 if (msg.has_input_data()) { | 74 if (msg.has_input_data()) { |
74 // Fixed interface processing. | 75 // Fixed interface processing. |
75 // Verify interface invariance. | 76 // Verify interface invariance. |
76 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || | 77 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || |
77 interface_used_ == InterfaceType::kNotSpecified); | 78 interface_used_ == InterfaceType::kNotSpecified); |
78 interface_used_ = InterfaceType::kFixedInterface; | 79 interface_used_ = InterfaceType::kFixedInterface; |
79 | 80 |
80 // Populate input buffer. | 81 // Populate input buffer. |
81 RTC_CHECK_EQ(sizeof(fwd_frame_.data_[0]) * fwd_frame_.samples_per_channel_ * | 82 RTC_CHECK_EQ(sizeof(fwd_frame_.data_[0]) * fwd_frame_.samples_per_channel_ * |
82 fwd_frame_.num_channels_, | 83 fwd_frame_.num_channels_, |
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151 if (!settings_.use_ts) { | 152 if (!settings_.use_ts) { |
152 if (msg.has_keypress()) { | 153 if (msg.has_keypress()) { |
153 ap_->set_stream_key_pressed(msg.keypress()); | 154 ap_->set_stream_key_pressed(msg.keypress()); |
154 } | 155 } |
155 } else { | 156 } else { |
156 ap_->set_stream_key_pressed(*settings_.use_ts); | 157 ap_->set_stream_key_pressed(*settings_.use_ts); |
157 } | 158 } |
158 | 159 |
159 // TODO(peah): Add support for controlling the analog level via the | 160 // TODO(peah): Add support for controlling the analog level via the |
160 // command-line. | 161 // command-line. |
161 if (msg.has_level()) { | 162 RTC_CHECK(msg.has_level()); // Level is always logged in AEC dumps. |
peah-webrtc
2017/05/02 21:27:32
It is probably fine, but I haven't checked the pro
| |
162 RTC_CHECK_EQ(AudioProcessing::kNoError, | 163 // When the analog gain is simulated, use the gain suggested by AGC instead of |
163 ap_->gain_control()->set_stream_analog_level(msg.level())); | 164 // that stored in the AEC dump. |
164 *set_stream_analog_level_called = true; | 165 RTC_CHECK_EQ(AudioProcessing::kNoError, |
165 } else { | 166 ap_->gain_control()->set_stream_analog_level( |
peah-webrtc
2017/05/02 21:27:32
What about putting the set_stream_analog_level cal
| |
166 *set_stream_analog_level_called = false; | 167 settings_.simulate_mic_gain ? |
167 } | 168 last_specified_microphone_level_ : msg.level())); |
169 // TODO(aleloi): If settings_.simulate_mic_gain, set undo level to | |
peah-webrtc
2017/05/02 21:27:33
It would be good to have FakeRecordingDevice as pa
| |
170 // |msg.level()| via FakeRecordingDevice::NotifyAudioDeviceLevel(). | |
168 } | 171 } |
169 | 172 |
170 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( | 173 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( |
171 const webrtc::audioproc::Stream& msg) { | 174 const webrtc::audioproc::Stream& msg) { |
172 if (bitexact_output_) { | 175 if (bitexact_output_) { |
173 if (interface_used_ == InterfaceType::kFixedInterface) { | 176 if (interface_used_ == InterfaceType::kFixedInterface) { |
174 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); | 177 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); |
175 } else { | 178 } else { |
176 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); | 179 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); |
177 } | 180 } |
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555 } | 558 } |
556 | 559 |
557 SetupBuffersConfigsOutputs( | 560 SetupBuffersConfigsOutputs( |
558 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), | 561 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), |
559 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, | 562 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, |
560 msg.num_reverse_channels(), num_reverse_output_channels); | 563 msg.num_reverse_channels(), num_reverse_output_channels); |
561 } | 564 } |
562 | 565 |
563 void AecDumpBasedSimulator::HandleMessage( | 566 void AecDumpBasedSimulator::HandleMessage( |
564 const webrtc::audioproc::Stream& msg) { | 567 const webrtc::audioproc::Stream& msg) { |
565 bool set_stream_analog_level_called = false; | 568 PrepareProcessStreamCall(msg); |
566 PrepareProcessStreamCall(msg, &set_stream_analog_level_called); | |
567 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); | 569 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); |
568 if (set_stream_analog_level_called) { | 570 if (settings_.simulate_mic_gain) { |
571 // Store analog level for the next analyzed frame. | |
572 last_specified_microphone_level_ = | |
573 ap_->gain_control()->stream_analog_level(); | |
peah-webrtc
2017/05/02 21:27:32
What about putting the stream_analog_level call in
| |
574 // TODO(aleloi): Set the returned value into a FakeRecordingDevice instance | |
575 // via FakeRecordingDevice::set_analog_level() instead of using | |
576 // last_specified_microphone_level_. | |
577 } else { | |
569 // Call stream analog level to ensure that any side-effects are triggered. | 578 // Call stream analog level to ensure that any side-effects are triggered. |
570 (void)ap_->gain_control()->stream_analog_level(); | 579 (void)ap_->gain_control()->stream_analog_level(); |
571 } | 580 } |
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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