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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 "webrtc/modules/audio_processing/gain_control_impl.h" | 11 #include "webrtc/modules/audio_processing/gain_control_impl.h" |
12 | 12 |
13 #include "webrtc/base/constructormagic.h" | 13 #include "webrtc/base/constructormagic.h" |
14 #include "webrtc/base/optional.h" | 14 #include "webrtc/base/optional.h" |
15 #include "webrtc/modules/audio_processing/audio_buffer.h" | 15 #include "webrtc/modules/audio_processing/audio_buffer.h" |
16 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h" | 16 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h" |
17 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | |
17 | 18 |
18 namespace webrtc { | 19 namespace webrtc { |
19 | 20 |
20 typedef void Handle; | 21 typedef void Handle; |
21 | 22 |
22 namespace { | 23 namespace { |
23 int16_t MapSetting(GainControl::Mode mode) { | 24 int16_t MapSetting(GainControl::Mode mode) { |
24 switch (mode) { | 25 switch (mode) { |
25 case GainControl::kAdaptiveAnalog: | 26 case GainControl::kAdaptiveAnalog: |
26 return kAgcModeAdaptiveAnalog; | 27 return kAgcModeAdaptiveAnalog; |
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77 | 78 |
78 private: | 79 private: |
79 Handle* state_; | 80 Handle* state_; |
80 // TODO(peah): Remove the optional once the initialization is moved into the | 81 // TODO(peah): Remove the optional once the initialization is moved into the |
81 // ctor. | 82 // ctor. |
82 rtc::Optional<int> capture_level_; | 83 rtc::Optional<int> capture_level_; |
83 | 84 |
84 RTC_DISALLOW_COPY_AND_ASSIGN(GainController); | 85 RTC_DISALLOW_COPY_AND_ASSIGN(GainController); |
85 }; | 86 }; |
86 | 87 |
88 int GainControlImpl::instance_counter_ = 0; | |
89 | |
87 GainControlImpl::GainControlImpl(rtc::CriticalSection* crit_render, | 90 GainControlImpl::GainControlImpl(rtc::CriticalSection* crit_render, |
88 rtc::CriticalSection* crit_capture) | 91 rtc::CriticalSection* crit_capture) |
89 : crit_render_(crit_render), | 92 : crit_render_(crit_render), |
90 crit_capture_(crit_capture), | 93 crit_capture_(crit_capture), |
94 data_dumper_(new ApmDataDumper(instance_counter_)), | |
ivoc
2016/10/28 09:20:10
The experimental AGC and the regular AGC could end
peah-webrtc
2016/10/28 09:43:41
They are not in practice active at the same time.
| |
91 mode_(kAdaptiveAnalog), | 95 mode_(kAdaptiveAnalog), |
92 minimum_capture_level_(0), | 96 minimum_capture_level_(0), |
93 maximum_capture_level_(255), | 97 maximum_capture_level_(255), |
94 limiter_enabled_(true), | 98 limiter_enabled_(true), |
95 target_level_dbfs_(3), | 99 target_level_dbfs_(3), |
96 compression_gain_db_(9), | 100 compression_gain_db_(9), |
97 analog_capture_level_(0), | 101 analog_capture_level_(0), |
98 was_analog_level_set_(false), | 102 was_analog_level_set_(false), |
99 stream_is_saturated_(false) { | 103 stream_is_saturated_(false) { |
100 RTC_DCHECK(crit_render); | 104 RTC_DCHECK(crit_render); |
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232 } | 236 } |
233 | 237 |
234 int GainControlImpl::compression_gain_db() const { | 238 int GainControlImpl::compression_gain_db() const { |
235 rtc::CritScope cs(crit_capture_); | 239 rtc::CritScope cs(crit_capture_); |
236 return compression_gain_db_; | 240 return compression_gain_db_; |
237 } | 241 } |
238 | 242 |
239 // TODO(ajm): ensure this is called under kAdaptiveAnalog. | 243 // TODO(ajm): ensure this is called under kAdaptiveAnalog. |
240 int GainControlImpl::set_stream_analog_level(int level) { | 244 int GainControlImpl::set_stream_analog_level(int level) { |
241 rtc::CritScope cs(crit_capture_); | 245 rtc::CritScope cs(crit_capture_); |
246 data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level); | |
ivoc
2016/10/28 09:20:10
Shouldn't there be a way to enable/disable this lo
peah-webrtc
2016/10/28 09:43:41
There is. What happens is that this method call ex
| |
242 | 247 |
243 was_analog_level_set_ = true; | 248 was_analog_level_set_ = true; |
244 if (level < minimum_capture_level_ || level > maximum_capture_level_) { | 249 if (level < minimum_capture_level_ || level > maximum_capture_level_) { |
245 return AudioProcessing::kBadParameterError; | 250 return AudioProcessing::kBadParameterError; |
246 } | 251 } |
247 analog_capture_level_ = level; | 252 analog_capture_level_ = level; |
248 | 253 |
249 return AudioProcessing::kNoError; | 254 return AudioProcessing::kNoError; |
250 } | 255 } |
251 | 256 |
252 int GainControlImpl::stream_analog_level() { | 257 int GainControlImpl::stream_analog_level() { |
253 rtc::CritScope cs(crit_capture_); | 258 rtc::CritScope cs(crit_capture_); |
259 data_dumper_->DumpRaw("gain_control_stream_analog_level", 1, | |
260 &analog_capture_level_); | |
254 // TODO(ajm): enable this assertion? | 261 // TODO(ajm): enable this assertion? |
255 //RTC_DCHECK_EQ(kAdaptiveAnalog, mode_); | 262 //RTC_DCHECK_EQ(kAdaptiveAnalog, mode_); |
256 | 263 |
257 return analog_capture_level_; | 264 return analog_capture_level_; |
258 } | 265 } |
259 | 266 |
260 int GainControlImpl::Enable(bool enable) { | 267 int GainControlImpl::Enable(bool enable) { |
261 rtc::CritScope cs_render(crit_render_); | 268 rtc::CritScope cs_render(crit_render_); |
262 rtc::CritScope cs_capture(crit_capture_); | 269 rtc::CritScope cs_capture(crit_capture_); |
263 if (enable && !enabled_) { | 270 if (enable && !enabled_) { |
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378 } | 385 } |
379 | 386 |
380 bool GainControlImpl::is_limiter_enabled() const { | 387 bool GainControlImpl::is_limiter_enabled() const { |
381 rtc::CritScope cs(crit_capture_); | 388 rtc::CritScope cs(crit_capture_); |
382 return limiter_enabled_; | 389 return limiter_enabled_; |
383 } | 390 } |
384 | 391 |
385 void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) { | 392 void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) { |
386 rtc::CritScope cs_render(crit_render_); | 393 rtc::CritScope cs_render(crit_render_); |
387 rtc::CritScope cs_capture(crit_capture_); | 394 rtc::CritScope cs_capture(crit_capture_); |
395 data_dumper_->InitiateNewSetOfRecordings(); | |
388 | 396 |
389 num_proc_channels_ = rtc::Optional<size_t>(num_proc_channels); | 397 num_proc_channels_ = rtc::Optional<size_t>(num_proc_channels); |
390 sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz); | 398 sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz); |
391 | 399 |
392 if (!enabled_) { | 400 if (!enabled_) { |
393 return; | 401 return; |
394 } | 402 } |
395 | 403 |
396 gain_controllers_.resize(*num_proc_channels_); | 404 gain_controllers_.resize(*num_proc_channels_); |
397 for (auto& gain_controller : gain_controllers_) { | 405 for (auto& gain_controller : gain_controllers_) { |
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422 for (auto& gain_controller : gain_controllers_) { | 430 for (auto& gain_controller : gain_controllers_) { |
423 const int handle_error = | 431 const int handle_error = |
424 WebRtcAgc_set_config(gain_controller->state(), config); | 432 WebRtcAgc_set_config(gain_controller->state(), config); |
425 if (handle_error != AudioProcessing::kNoError) { | 433 if (handle_error != AudioProcessing::kNoError) { |
426 error = handle_error; | 434 error = handle_error; |
427 } | 435 } |
428 } | 436 } |
429 return error; | 437 return error; |
430 } | 438 } |
431 } // namespace webrtc | 439 } // namespace webrtc |
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