Chromium Code Reviews| Index: webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h |
| diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..d1123dff0bd5934a80f9ae2f96fbb6d3a6329e0c |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h |
| @@ -0,0 +1,53 @@ |
| +/* |
| + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| + * |
| + * Use of this source code is governed by a BSD-style license |
| + * that can be found in the LICENSE file in the root of the source |
| + * tree. An additional intellectual property rights grant can be found |
| + * in the file PATENTS. All contributing project authors may |
| + * be found in the AUTHORS file in the root of the source tree. |
| + */ |
| + |
| +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_ |
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_ |
| + |
| +#include <memory> |
| +#include <string> |
| + |
| +#include "webrtc/base/platform_file.h" |
| +#include "webrtc/modules/audio_processing/include/aec_dump.h" |
| + |
| +namespace rtc { |
| +class TaskQueue; |
| +} // namespace rtc |
| + |
| +namespace webrtc { |
| + |
| +class AecDumpFactory { |
| + public: |
| + // TODO(aleloi): update comments to new creation scheme. |
| + // If called when a recording is active, that file is closed, and a |
| + // new file is opened. Messages waiting to be written asynchronously |
| + // to the old file may be lost. Returns true iff opening file for |
| + // writing succeeded. |
| + |
| + // Closes associated file. Messages waiting to be written to file |
|
peah-webrtc
2017/05/16 06:30:37
What does this comment refer to?
aleloi
2017/05/16 20:10:16
Now updated. I've went through the other comments
|
| + // asynchronously may be lost. This method is safe to call when no |
| + // recording is active. A recording does not have to be closed |
| + // manually with this method; instead the AecDump instance may be |
| + // destroyed. |
| + |
| + static std::unique_ptr<AecDump> Create(rtc::PlatformFile file, |
| + int64_t max_log_size_bytes, |
| + rtc::TaskQueue* worker_queue); |
| + static std::unique_ptr<AecDump> Create(std::string file_name, |
| + int64_t max_log_size_bytes, |
| + rtc::TaskQueue* worker_queue); |
| + static std::unique_ptr<AecDump> Create(FILE* handle, |
| + int64_t max_log_size_bytes, |
| + rtc::TaskQueue* worker_queue); |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_ |