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Unified Diff: webrtc/modules/utility/source/file_recorder_impl.h

Issue 2037623002: Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove0
Patch Set: GN build fix Created 4 years, 6 months ago
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Index: webrtc/modules/utility/source/file_recorder_impl.h
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
deleted file mode 100644
index a9dd3a88633afe44bbd059cba056bcdad58dac61..0000000000000000000000000000000000000000
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-// This file contains a class that can write audio to file in
-// multiple file formats. The unencoded input data is written to file in the
-// encoded format specified.
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-
-#include <list>
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/media_file/media_file.h"
-#include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/include/file_recorder.h"
-#include "webrtc/modules/utility/source/coder.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-// The largest decoded frame size in samples (60ms with 32kHz sample rate).
-enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
-enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
-enum { kMaxAudioBufferQueueLength = 100 };
-
-class CriticalSectionWrapper;
-
-class FileRecorderImpl : public FileRecorder
-{
-public:
- FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
- virtual ~FileRecorderImpl();
-
- // FileRecorder functions.
- int32_t RegisterModuleFileCallback(FileCallback* callback) override;
- FileFormats RecordingFileFormat() const override;
- int32_t StartRecordingAudioFile(
- const char* fileName,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) override;
- int32_t StartRecordingAudioFile(
- OutStream& destStream,
- const CodecInst& codecInst,
- uint32_t notificationTimeMs) override;
- int32_t StopRecording() override;
- bool IsRecording() const override;
- int32_t codec_info(CodecInst& codecInst) const override;
- int32_t RecordAudioToFile(const AudioFrame& frame) override;
-
-protected:
- int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
- size_t bufferLength);
-
- int32_t SetUpAudioEncoder();
-
- uint32_t _instanceID;
- FileFormats _fileFormat;
- MediaFile* _moduleFile;
-
-private:
- CodecInst codec_info_;
- int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
- AudioCoder _audioEncoder;
- Resampler _audioResampler;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
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