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Side by Side Diff: webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

Issue 2834643002: audioproc_f with simulated mic analog gain (Closed)
Patch Set: AEC dump + fake rec device bugfix Created 3 years, 3 months ago
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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 <iostream> 11 #include <iostream>
12 12
13 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h" 13 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h"
14 14
15 #include "webrtc/modules/audio_processing/test/protobuf_utils.h" 15 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
16 #include "webrtc/rtc_base/checks.h" 16 #include "webrtc/rtc_base/checks.h"
17 #include "webrtc/rtc_base/logging.h"
17 #include "webrtc/test/testsupport/trace_to_stderr.h" 18 #include "webrtc/test/testsupport/trace_to_stderr.h"
18 19
19 namespace webrtc { 20 namespace webrtc {
20 namespace test { 21 namespace test {
21 namespace { 22 namespace {
22 23
23 // Verify output bitexactness for the fixed interface. 24 // Verify output bitexactness for the fixed interface.
24 // TODO(peah): Check whether it would make sense to add a threshold 25 // TODO(peah): Check whether it would make sense to add a threshold
25 // to use for checking the bitexactness in a soft manner. 26 // to use for checking the bitexactness in a soft manner.
26 bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg, 27 bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
(...skipping 30 matching lines...) Expand all
57 } 58 }
58 } 59 }
59 } 60 }
60 } 61 }
61 return true; 62 return true;
62 } 63 }
63 64
64 } // namespace 65 } // namespace
65 66
66 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) 67 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings)
67 : AudioProcessingSimulator(settings) {} 68 : AudioProcessingSimulator(settings) {
69 if (settings_.simulate_mic_gain)
70 LOG(LS_VERBOSE) << "Simulating analog mic gain using AEC dump as input";
71 }
68 72
69 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; 73 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
70 74
71 void AecDumpBasedSimulator::PrepareProcessStreamCall( 75 void AecDumpBasedSimulator::PrepareProcessStreamCall(
72 const webrtc::audioproc::Stream& msg, 76 const webrtc::audioproc::Stream& msg) {
73 bool* set_stream_analog_level_called) {
74 if (msg.has_input_data()) { 77 if (msg.has_input_data()) {
75 // Fixed interface processing. 78 // Fixed interface processing.
76 // Verify interface invariance. 79 // Verify interface invariance.
77 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface || 80 RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
78 interface_used_ == InterfaceType::kNotSpecified); 81 interface_used_ == InterfaceType::kNotSpecified);
79 interface_used_ = InterfaceType::kFixedInterface; 82 interface_used_ = InterfaceType::kFixedInterface;
80 83
81 // Populate input buffer. 84 // Populate input buffer.
82 RTC_CHECK_EQ(sizeof(*fwd_frame_.data()) * fwd_frame_.samples_per_channel_ * 85 RTC_CHECK_EQ(sizeof(*fwd_frame_.data()) * fwd_frame_.samples_per_channel_ *
83 fwd_frame_.num_channels_, 86 fwd_frame_.num_channels_,
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152 } 155 }
153 156
154 if (!settings_.use_ts) { 157 if (!settings_.use_ts) {
155 if (msg.has_keypress()) { 158 if (msg.has_keypress()) {
156 ap_->set_stream_key_pressed(msg.keypress()); 159 ap_->set_stream_key_pressed(msg.keypress());
157 } 160 }
158 } else { 161 } else {
159 ap_->set_stream_key_pressed(*settings_.use_ts); 162 ap_->set_stream_key_pressed(*settings_.use_ts);
160 } 163 }
161 164
162 // TODO(peah): Add support for controlling the analog level via the 165 // Level is always logged in AEC dumps.
163 // command-line. 166 RTC_CHECK(msg.has_level());
164 if (msg.has_level()) { 167 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 } 168 }
172 169
173 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness( 170 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
174 const webrtc::audioproc::Stream& msg) { 171 const webrtc::audioproc::Stream& msg) {
175 if (bitexact_output_) { 172 if (bitexact_output_) {
176 if (interface_used_ == InterfaceType::kFixedInterface) { 173 if (interface_used_ == InterfaceType::kFixedInterface) {
177 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_); 174 bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
178 } else { 175 } else {
179 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_); 176 bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
180 } 177 }
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558 } 555 }
559 556
560 SetupBuffersConfigsOutputs( 557 SetupBuffersConfigsOutputs(
561 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(), 558 msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(),
562 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels, 559 reverse_output_sample_rate, msg.num_input_channels(), num_output_channels,
563 msg.num_reverse_channels(), num_reverse_output_channels); 560 msg.num_reverse_channels(), num_reverse_output_channels);
564 } 561 }
565 562
566 void AecDumpBasedSimulator::HandleMessage( 563 void AecDumpBasedSimulator::HandleMessage(
567 const webrtc::audioproc::Stream& msg) { 564 const webrtc::audioproc::Stream& msg) {
568 bool set_stream_analog_level_called = false; 565 PrepareProcessStreamCall(msg);
569 PrepareProcessStreamCall(msg, &set_stream_analog_level_called);
570 ProcessStream(interface_used_ == InterfaceType::kFixedInterface); 566 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); 567 VerifyProcessStreamBitExactness(msg);
577 } 568 }
578 569
579 void AecDumpBasedSimulator::HandleMessage( 570 void AecDumpBasedSimulator::HandleMessage(
580 const webrtc::audioproc::ReverseStream& msg) { 571 const webrtc::audioproc::ReverseStream& msg) {
581 PrepareReverseProcessStreamCall(msg); 572 PrepareReverseProcessStreamCall(msg);
582 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface); 573 ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
583 } 574 }
584 575
585 } // namespace test 576 } // namespace test
586 } // namespace webrtc 577 } // namespace webrtc
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