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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 |
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233 } | 233 } |
234 | 234 |
235 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive() | 235 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive() |
236 const { | 236 const { |
237 return low_cut_filter_enabled_ || echo_canceller_enabled_ || | 237 return low_cut_filter_enabled_ || echo_canceller_enabled_ || |
238 mobile_echo_controller_enabled_ || noise_suppressor_enabled_ || | 238 mobile_echo_controller_enabled_ || noise_suppressor_enabled_ || |
239 beamformer_enabled_ || adaptive_gain_controller_enabled_ || | 239 beamformer_enabled_ || adaptive_gain_controller_enabled_ || |
240 echo_canceller3_enabled_; | 240 echo_canceller3_enabled_; |
241 } | 241 } |
242 | 242 |
| 243 bool AudioProcessingImpl::ApmSubmoduleStates::CaptureFullBandProcessingActive() |
| 244 const { |
| 245 return level_controller_enabled_; |
| 246 } |
| 247 |
243 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive() | 248 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive() |
244 const { | 249 const { |
245 return RenderMultiBandProcessingActive() || echo_canceller_enabled_ || | 250 return RenderMultiBandProcessingActive() || echo_canceller_enabled_ || |
246 mobile_echo_controller_enabled_ || adaptive_gain_controller_enabled_ || | 251 mobile_echo_controller_enabled_ || adaptive_gain_controller_enabled_ || |
247 echo_canceller3_enabled_; | 252 echo_canceller3_enabled_; |
248 } | 253 } |
249 | 254 |
250 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandProcessingActive() | 255 bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandProcessingActive() |
251 const { | 256 const { |
252 #if WEBRTC_INTELLIGIBILITY_ENHANCER | 257 #if WEBRTC_INTELLIGIBILITY_ENHANCER |
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1126 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); | 1131 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
1127 const size_t data_size = | 1132 const size_t data_size = |
1128 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; | 1133 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
1129 msg->set_input_data(frame->data_, data_size); | 1134 msg->set_input_data(frame->data_, data_size); |
1130 } | 1135 } |
1131 #endif | 1136 #endif |
1132 | 1137 |
1133 capture_.capture_audio->DeinterleaveFrom(frame); | 1138 capture_.capture_audio->DeinterleaveFrom(frame); |
1134 RETURN_ON_ERR(ProcessCaptureStreamLocked()); | 1139 RETURN_ON_ERR(ProcessCaptureStreamLocked()); |
1135 capture_.capture_audio->InterleaveTo( | 1140 capture_.capture_audio->InterleaveTo( |
1136 frame, submodule_states_.CaptureMultiBandProcessingActive()); | 1141 frame, submodule_states_.CaptureMultiBandProcessingActive() || |
| 1142 submodule_states_.CaptureFullBandProcessingActive()); |
1137 | 1143 |
1138 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 1144 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
1139 if (debug_dump_.debug_file->is_open()) { | 1145 if (debug_dump_.debug_file->is_open()) { |
1140 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); | 1146 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
1141 const size_t data_size = | 1147 const size_t data_size = |
1142 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; | 1148 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
1143 msg->set_output_data(frame->data_, data_size); | 1149 msg->set_output_data(frame->data_, data_size); |
1144 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), | 1150 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), |
1145 &debug_dump_.num_bytes_left_for_log_, | 1151 &debug_dump_.num_bytes_left_for_log_, |
1146 &crit_debug_, &debug_dump_.capture)); | 1152 &crit_debug_, &debug_dump_.capture)); |
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2064 previous_agc_level(0), | 2070 previous_agc_level(0), |
2065 echo_path_gain_change(false) {} | 2071 echo_path_gain_change(false) {} |
2066 | 2072 |
2067 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; | 2073 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; |
2068 | 2074 |
2069 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; | 2075 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; |
2070 | 2076 |
2071 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; | 2077 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; |
2072 | 2078 |
2073 } // namespace webrtc | 2079 } // namespace webrtc |
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