1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_processing/test/debug_dump_replayer.h"
12
13 #include <string>
14
15 #include "absl/strings/string_view.h"
16 #include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
17 #include "modules/audio_processing/test/protobuf_utils.h"
18 #include "modules/audio_processing/test/runtime_setting_util.h"
19 #include "rtc_base/checks.h"
20
21 namespace webrtc {
22 namespace test {
23
24 namespace {
25
MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>> * buffer,const StreamConfig & config)26 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
27 const StreamConfig& config) {
28 auto& buffer_ref = *buffer;
29 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
30 buffer_ref->num_channels() != config.num_channels()) {
31 buffer_ref.reset(
32 new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
33 }
34 }
35
36 } // namespace
37
DebugDumpReplayer()38 DebugDumpReplayer::DebugDumpReplayer()
39 : input_(nullptr), // will be created upon usage.
40 reverse_(nullptr),
41 output_(nullptr),
42 apm_(nullptr),
43 debug_file_(nullptr) {}
44
~DebugDumpReplayer()45 DebugDumpReplayer::~DebugDumpReplayer() {
46 if (debug_file_)
47 fclose(debug_file_);
48 }
49
SetDumpFile(absl::string_view filename)50 bool DebugDumpReplayer::SetDumpFile(absl::string_view filename) {
51 debug_file_ = fopen(std::string(filename).c_str(), "rb");
52 LoadNextMessage();
53 return debug_file_;
54 }
55
56 // Get next event that has not run.
GetNextEvent() const57 absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
58 if (!has_next_event_)
59 return absl::nullopt;
60 else
61 return next_event_;
62 }
63
64 // Run the next event. Returns the event type.
RunNextEvent()65 bool DebugDumpReplayer::RunNextEvent() {
66 if (!has_next_event_)
67 return false;
68 switch (next_event_.type()) {
69 case audioproc::Event::INIT:
70 OnInitEvent(next_event_.init());
71 break;
72 case audioproc::Event::STREAM:
73 OnStreamEvent(next_event_.stream());
74 break;
75 case audioproc::Event::REVERSE_STREAM:
76 OnReverseStreamEvent(next_event_.reverse_stream());
77 break;
78 case audioproc::Event::CONFIG:
79 OnConfigEvent(next_event_.config());
80 break;
81 case audioproc::Event::RUNTIME_SETTING:
82 OnRuntimeSettingEvent(next_event_.runtime_setting());
83 break;
84 case audioproc::Event::UNKNOWN_EVENT:
85 // We do not expect to receive UNKNOWN event.
86 RTC_CHECK_NOTREACHED();
87 }
88 LoadNextMessage();
89 return true;
90 }
91
GetOutput() const92 const ChannelBuffer<float>* DebugDumpReplayer::GetOutput() const {
93 return output_.get();
94 }
95
GetOutputConfig() const96 StreamConfig DebugDumpReplayer::GetOutputConfig() const {
97 return output_config_;
98 }
99
100 // OnInitEvent reset the input/output/reserve channel format.
OnInitEvent(const audioproc::Init & msg)101 void DebugDumpReplayer::OnInitEvent(const audioproc::Init& msg) {
102 RTC_CHECK(msg.has_num_input_channels());
103 RTC_CHECK(msg.has_output_sample_rate());
104 RTC_CHECK(msg.has_num_output_channels());
105 RTC_CHECK(msg.has_reverse_sample_rate());
106 RTC_CHECK(msg.has_num_reverse_channels());
107
108 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
109 output_config_ =
110 StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
111 reverse_config_ =
112 StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
113
114 MaybeResetBuffer(&input_, input_config_);
115 MaybeResetBuffer(&output_, output_config_);
116 MaybeResetBuffer(&reverse_, reverse_config_);
117 }
118
119 // OnStreamEvent replays an input signal and verifies the output.
OnStreamEvent(const audioproc::Stream & msg)120 void DebugDumpReplayer::OnStreamEvent(const audioproc::Stream& msg) {
121 // APM should have been created.
122 RTC_CHECK(apm_.get());
123
124 if (msg.has_applied_input_volume()) {
125 apm_->set_stream_analog_level(msg.applied_input_volume());
126 }
127 RTC_CHECK_EQ(AudioProcessing::kNoError,
128 apm_->set_stream_delay_ms(msg.delay()));
129
130 if (msg.has_keypress()) {
131 apm_->set_stream_key_pressed(msg.keypress());
132 } else {
133 apm_->set_stream_key_pressed(true);
134 }
135
136 RTC_CHECK_EQ(input_config_.num_channels(),
137 static_cast<size_t>(msg.input_channel_size()));
138 RTC_CHECK_EQ(input_config_.num_frames() * sizeof(float),
139 msg.input_channel(0).size());
140
141 for (int i = 0; i < msg.input_channel_size(); ++i) {
142 memcpy(input_->channels()[i], msg.input_channel(i).data(),
143 msg.input_channel(i).size());
144 }
145
146 RTC_CHECK_EQ(AudioProcessing::kNoError,
147 apm_->ProcessStream(input_->channels(), input_config_,
148 output_config_, output_->channels()));
149 }
150
OnReverseStreamEvent(const audioproc::ReverseStream & msg)151 void DebugDumpReplayer::OnReverseStreamEvent(
152 const audioproc::ReverseStream& msg) {
153 // APM should have been created.
154 RTC_CHECK(apm_.get());
155
156 RTC_CHECK_GT(msg.channel_size(), 0);
157 RTC_CHECK_EQ(reverse_config_.num_channels(),
158 static_cast<size_t>(msg.channel_size()));
159 RTC_CHECK_EQ(reverse_config_.num_frames() * sizeof(float),
160 msg.channel(0).size());
161
162 for (int i = 0; i < msg.channel_size(); ++i) {
163 memcpy(reverse_->channels()[i], msg.channel(i).data(),
164 msg.channel(i).size());
165 }
166
167 RTC_CHECK_EQ(
168 AudioProcessing::kNoError,
169 apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
170 reverse_config_, reverse_->channels()));
171 }
172
OnConfigEvent(const audioproc::Config & msg)173 void DebugDumpReplayer::OnConfigEvent(const audioproc::Config& msg) {
174 MaybeRecreateApm(msg);
175 ConfigureApm(msg);
176 }
177
OnRuntimeSettingEvent(const audioproc::RuntimeSetting & msg)178 void DebugDumpReplayer::OnRuntimeSettingEvent(
179 const audioproc::RuntimeSetting& msg) {
180 RTC_CHECK(apm_.get());
181 ReplayRuntimeSetting(apm_.get(), msg);
182 }
183
MaybeRecreateApm(const audioproc::Config & msg)184 void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
185 // These configurations cannot be changed on the fly.
186 RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
187 RTC_CHECK(msg.has_aec_extended_filter_enabled());
188
189 // We only create APM once, since changes on these fields should not
190 // happen in current implementation.
191 if (!apm_.get()) {
192 apm_ = AudioProcessingBuilderForTesting().Create();
193 }
194 }
195
ConfigureApm(const audioproc::Config & msg)196 void DebugDumpReplayer::ConfigureApm(const audioproc::Config& msg) {
197 AudioProcessing::Config apm_config;
198
199 // AEC2/AECM configs.
200 RTC_CHECK(msg.has_aec_enabled());
201 RTC_CHECK(msg.has_aecm_enabled());
202 apm_config.echo_canceller.enabled = msg.aec_enabled() || msg.aecm_enabled();
203 apm_config.echo_canceller.mobile_mode = msg.aecm_enabled();
204
205 // HPF configs.
206 RTC_CHECK(msg.has_hpf_enabled());
207 apm_config.high_pass_filter.enabled = msg.hpf_enabled();
208
209 // Preamp configs.
210 RTC_CHECK(msg.has_pre_amplifier_enabled());
211 apm_config.pre_amplifier.enabled = msg.pre_amplifier_enabled();
212 apm_config.pre_amplifier.fixed_gain_factor =
213 msg.pre_amplifier_fixed_gain_factor();
214
215 // NS configs.
216 RTC_CHECK(msg.has_ns_enabled());
217 RTC_CHECK(msg.has_ns_level());
218 apm_config.noise_suppression.enabled = msg.ns_enabled();
219 apm_config.noise_suppression.level =
220 static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
221 msg.ns_level());
222
223 // TS configs.
224 RTC_CHECK(msg.has_transient_suppression_enabled());
225 apm_config.transient_suppression.enabled =
226 msg.transient_suppression_enabled();
227
228 // AGC configs.
229 RTC_CHECK(msg.has_agc_enabled());
230 RTC_CHECK(msg.has_agc_mode());
231 RTC_CHECK(msg.has_agc_limiter_enabled());
232 apm_config.gain_controller1.enabled = msg.agc_enabled();
233 apm_config.gain_controller1.mode =
234 static_cast<AudioProcessing::Config::GainController1::Mode>(
235 msg.agc_mode());
236 apm_config.gain_controller1.enable_limiter = msg.agc_limiter_enabled();
237 RTC_CHECK(msg.has_noise_robust_agc_enabled());
238 apm_config.gain_controller1.analog_gain_controller.enabled =
239 msg.noise_robust_agc_enabled();
240
241 apm_->ApplyConfig(apm_config);
242 }
243
LoadNextMessage()244 void DebugDumpReplayer::LoadNextMessage() {
245 has_next_event_ =
246 debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
247 }
248
249 } // namespace test
250 } // namespace webrtc
251