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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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1153 } | 1153 } |
1154 | 1154 |
1155 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 1155 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
1156 if (debug_dump_.debug_file->is_open()) { | 1156 if (debug_dump_.debug_file->is_open()) { |
1157 RETURN_ON_ERR(WriteConfigMessage(false)); | 1157 RETURN_ON_ERR(WriteConfigMessage(false)); |
1158 | 1158 |
1159 debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM); | 1159 debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM); |
1160 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); | 1160 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
1161 const size_t data_size = | 1161 const size_t data_size = |
1162 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; | 1162 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
1163 msg->set_input_data(frame->data_, data_size); | 1163 msg->set_input_data(frame->data(), data_size); |
1164 } | 1164 } |
1165 #endif | 1165 #endif |
1166 | 1166 |
1167 capture_.capture_audio->DeinterleaveFrom(frame); | 1167 capture_.capture_audio->DeinterleaveFrom(frame); |
1168 RETURN_ON_ERR(ProcessCaptureStreamLocked()); | 1168 RETURN_ON_ERR(ProcessCaptureStreamLocked()); |
1169 capture_.capture_audio->InterleaveTo( | 1169 capture_.capture_audio->InterleaveTo( |
1170 frame, submodule_states_.CaptureMultiBandProcessingActive() || | 1170 frame, submodule_states_.CaptureMultiBandProcessingActive() || |
1171 submodule_states_.CaptureFullBandProcessingActive()); | 1171 submodule_states_.CaptureFullBandProcessingActive()); |
1172 | 1172 |
1173 if (aec_dump_) { | 1173 if (aec_dump_) { |
1174 RecordProcessedCaptureStream(*frame); | 1174 RecordProcessedCaptureStream(*frame); |
1175 } | 1175 } |
1176 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 1176 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
1177 if (debug_dump_.debug_file->is_open()) { | 1177 if (debug_dump_.debug_file->is_open()) { |
1178 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); | 1178 audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream(); |
1179 const size_t data_size = | 1179 const size_t data_size = |
1180 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; | 1180 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
1181 msg->set_output_data(frame->data_, data_size); | 1181 msg->set_output_data(frame->data(), data_size); |
1182 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), | 1182 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), |
1183 &debug_dump_.num_bytes_left_for_log_, | 1183 &debug_dump_.num_bytes_left_for_log_, |
1184 &crit_debug_, &debug_dump_.capture)); | 1184 &crit_debug_, &debug_dump_.capture)); |
1185 } | 1185 } |
1186 #endif | 1186 #endif |
1187 | 1187 |
1188 return kNoError; | 1188 return kNoError; |
1189 } | 1189 } |
1190 | 1190 |
1191 int AudioProcessingImpl::ProcessCaptureStreamLocked() { | 1191 int AudioProcessingImpl::ProcessCaptureStreamLocked() { |
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1507 return kBadDataLengthError; | 1507 return kBadDataLengthError; |
1508 } | 1508 } |
1509 | 1509 |
1510 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 1510 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
1511 if (debug_dump_.debug_file->is_open()) { | 1511 if (debug_dump_.debug_file->is_open()) { |
1512 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); | 1512 debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM); |
1513 audioproc::ReverseStream* msg = | 1513 audioproc::ReverseStream* msg = |
1514 debug_dump_.render.event_msg->mutable_reverse_stream(); | 1514 debug_dump_.render.event_msg->mutable_reverse_stream(); |
1515 const size_t data_size = | 1515 const size_t data_size = |
1516 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; | 1516 sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_; |
1517 msg->set_data(frame->data_, data_size); | 1517 msg->set_data(frame->data(), data_size); |
1518 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), | 1518 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), |
1519 &debug_dump_.num_bytes_left_for_log_, | 1519 &debug_dump_.num_bytes_left_for_log_, |
1520 &crit_debug_, &debug_dump_.render)); | 1520 &crit_debug_, &debug_dump_.render)); |
1521 } | 1521 } |
1522 #endif | 1522 #endif |
1523 if (aec_dump_) { | 1523 if (aec_dump_) { |
1524 aec_dump_->WriteRenderStreamMessage(*frame); | 1524 aec_dump_->WriteRenderStreamMessage(*frame); |
1525 } | 1525 } |
1526 | 1526 |
1527 render_.render_audio->DeinterleaveFrom(frame); | 1527 render_.render_audio->DeinterleaveFrom(frame); |
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2255 previous_agc_level(0), | 2255 previous_agc_level(0), |
2256 echo_path_gain_change(false) {} | 2256 echo_path_gain_change(false) {} |
2257 | 2257 |
2258 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; | 2258 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; |
2259 | 2259 |
2260 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; | 2260 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; |
2261 | 2261 |
2262 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; | 2262 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; |
2263 | 2263 |
2264 } // namespace webrtc | 2264 } // namespace webrtc |
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