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 |
new file mode 100644 |
index 0000000000000000000000000000000000000000..a9dd3a88633afe44bbd059cba056bcdad58dac61 |
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+++ b/webrtc/modules/utility/source/file_recorder_impl.h |
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+/* |
+ * 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_ |