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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> | |
peah-webrtc
2017/06/29 05:45:26
Why was algorithm and utility added to the include
AleBzk
2017/06/29 11:43:35
Good catch! Thanks!
| |
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" | |
peah-webrtc
2017/06/29 05:45:26
This include is not needed, right?
AleBzk
2017/06/29 11:43:35
Yet another good catch :)
| |
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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57 } | 61 } |
58 } | 62 } |
59 } | 63 } |
60 } | 64 } |
61 return true; | 65 return true; |
62 } | 66 } |
63 | 67 |
64 } // namespace | 68 } // namespace |
65 | 69 |
66 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) | 70 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) |
67 : AudioProcessingSimulator(settings) {} | 71 : AudioProcessingSimulator(settings) { |
72 if (settings_.simulate_mic_gain) { | |
73 LOG(LS_VERBOSE) << "Simulating analog mic gain using AEC dump as input " | |
74 << "(the unmodified mic gain level will be virtually restored)"; | |
75 } | |
76 } | |
68 | 77 |
69 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; | 78 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; |
70 | 79 |
71 void AecDumpBasedSimulator::PrepareProcessStreamCall( | 80 void AecDumpBasedSimulator::PrepareProcessStreamCall( |
72 const webrtc::audioproc::Stream& msg, | 81 const webrtc::audioproc::Stream& msg) { |
73 bool* set_stream_analog_level_called) { | |
74 if (msg.has_input_data()) { | 82 if (msg.has_input_data()) { |
75 // Fixed interface processing. | 83 // Fixed interface processing. |
76 // Verify interface invariance. | 84 // Verify interface invariance. |
77 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || | 85 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || |
78 interface_used_ == InterfaceType::kNotSpecified); | 86 interface_used_ == InterfaceType::kNotSpecified); |
79 interface_used_ = InterfaceType::kFixedInterface; | 87 interface_used_ = InterfaceType::kFixedInterface; |
80 | 88 |
81 // Populate input buffer. | 89 // Populate input buffer. |
82 RTC_CHECK_EQ(sizeof(*fwd_frame_.data()) * fwd_frame_.samples_per_channel_ * | 90 RTC_CHECK_EQ(sizeof(*fwd_frame_.data()) * fwd_frame_.samples_per_channel_ * |
83 fwd_frame_.num_channels_, | 91 fwd_frame_.num_channels_, |
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152 } | 160 } |
153 | 161 |
154 if (!settings_.use_ts) { | 162 if (!settings_.use_ts) { |
155 if (msg.has_keypress()) { | 163 if (msg.has_keypress()) { |
156 ap_->set_stream_key_pressed(msg.keypress()); | 164 ap_->set_stream_key_pressed(msg.keypress()); |
157 } | 165 } |
158 } else { | 166 } else { |
159 ap_->set_stream_key_pressed(*settings_.use_ts); | 167 ap_->set_stream_key_pressed(*settings_.use_ts); |
160 } | 168 } |
161 | 169 |
162 // TODO(peah): Add support for controlling the analog level via the | 170 // Level is always logged in AEC dumps. |
163 // command-line. | 171 RTC_CHECK(msg.has_level()); |
164 if (msg.has_level()) { | 172 aec_dump_mic_level_ = rtc::Optional<int>(msg.level()); |
165 RTC_CHECK_EQ(AudioProcessing::kNoError, | |
166 ap_->gain_control()->set_stream_analog_level(msg.level())); | |
167 *set_stream_analog_level_called = true; | |
168 } else { | |
169 *set_stream_analog_level_called = false; | |
170 } | |
171 } | 173 } |
172 | 174 |
173 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( | 175 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( |
174 const webrtc::audioproc::Stream& msg) { | 176 const webrtc::audioproc::Stream& msg) { |
175 if (bitexact_output_) { | 177 if (bitexact_output_) { |
176 if (interface_used_ == InterfaceType::kFixedInterface) { | 178 if (interface_used_ == InterfaceType::kFixedInterface) { |
177 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); | 179 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); |
178 } else { | 180 } else { |
179 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); | 181 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); |
180 } | 182 } |
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558 } | 560 } |
559 | 561 |
560 SetupBuffersConfigsOutputs( | 562 SetupBuffersConfigsOutputs( |
561 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), | 563 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), |
562 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, | 564 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, |
563 msg.num_reverse_channels(), num_reverse_output_channels); | 565 msg.num_reverse_channels(), num_reverse_output_channels); |
564 } | 566 } |
565 | 567 |
566 void AecDumpBasedSimulator::HandleMessage( | 568 void AecDumpBasedSimulator::HandleMessage( |
567 const webrtc::audioproc::Stream& msg) { | 569 const webrtc::audioproc::Stream& msg) { |
568 bool set_stream_analog_level_called = false; | 570 PrepareProcessStreamCall(msg); |
569 PrepareProcessStreamCall(msg, &set_stream_analog_level_called); | |
570 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); | 571 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); |
571 if (set_stream_analog_level_called) { | |
572 // Call stream analog level to ensure that any side-effects are triggered. | |
573 (void)ap_->gain_control()->stream_analog_level(); | |
574 } | |
575 | |
576 VerifyProcessStreamBitExactness(msg); | 572 VerifyProcessStreamBitExactness(msg); |
577 } | 573 } |
578 | 574 |
579 void AecDumpBasedSimulator::HandleMessage( | 575 void AecDumpBasedSimulator::HandleMessage( |
580 const webrtc::audioproc::ReverseStream& msg) { | 576 const webrtc::audioproc::ReverseStream& msg) { |
581 PrepareReverseProcessStreamCall(msg); | 577 PrepareReverseProcessStreamCall(msg); |
582 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); | 578 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); |
583 } | 579 } |
584 | 580 |
585 } // namespace test | 581 } // namespace test |
586 } // namespace webrtc | 582 } // namespace webrtc |
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