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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1878533002: Store reverse output rate and number of channels in APM protobuf recordings (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
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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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1385 msg->set_num_input_channels(static_cast<google::protobuf::int32>( 1385 msg->set_num_input_channels(static_cast<google::protobuf::int32>(
1386 formats_.api_format.input_stream().num_channels())); 1386 formats_.api_format.input_stream().num_channels()));
1387 msg->set_num_output_channels(static_cast<google::protobuf::int32>( 1387 msg->set_num_output_channels(static_cast<google::protobuf::int32>(
1388 formats_.api_format.output_stream().num_channels())); 1388 formats_.api_format.output_stream().num_channels()));
1389 msg->set_num_reverse_channels(static_cast<google::protobuf::int32>( 1389 msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
1390 formats_.api_format.reverse_input_stream().num_channels())); 1390 formats_.api_format.reverse_input_stream().num_channels()));
1391 msg->set_reverse_sample_rate( 1391 msg->set_reverse_sample_rate(
1392 formats_.api_format.reverse_input_stream().sample_rate_hz()); 1392 formats_.api_format.reverse_input_stream().sample_rate_hz());
1393 msg->set_output_sample_rate( 1393 msg->set_output_sample_rate(
1394 formats_.api_format.output_stream().sample_rate_hz()); 1394 formats_.api_format.output_stream().sample_rate_hz());
1395 // TODO(ekmeyerson): Add reverse output fields to 1395 msg->set_reverse_output_sample_rate(
1396 // debug_dump_.capture.event_msg. 1396 formats_.api_format.reverse_output_stream().sample_rate_hz());
1397 msg->set_num_reverse_output_channels(
1398 formats_.api_format.reverse_output_stream().num_channels());
1397 1399
1398 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 1400 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
1399 &debug_dump_.num_bytes_left_for_log_, 1401 &debug_dump_.num_bytes_left_for_log_,
1400 &crit_debug_, &debug_dump_.capture)); 1402 &crit_debug_, &debug_dump_.capture));
1401 return kNoError; 1403 return kNoError;
1402 } 1404 }
1403 1405
1404 int AudioProcessingImpl::WriteConfigMessage(bool forced) { 1406 int AudioProcessingImpl::WriteConfigMessage(bool forced) {
1405 audioproc::Config config; 1407 audioproc::Config config;
1406 1408
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1449 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config); 1451 debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
1450 1452
1451 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(), 1453 RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
1452 &debug_dump_.num_bytes_left_for_log_, 1454 &debug_dump_.num_bytes_left_for_log_,
1453 &crit_debug_, &debug_dump_.capture)); 1455 &crit_debug_, &debug_dump_.capture));
1454 return kNoError; 1456 return kNoError;
1455 } 1457 }
1456 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1458 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1457 1459
1458 } // namespace webrtc 1460 } // namespace webrtc
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