git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/utility/source/file_recorder_impl.h
+++ b/modules/utility/source/file_recorder_impl.h
@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2011 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 and/or video 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 "coder.h"
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "event_wrapper.h"
+#include "file_recorder.h"
+#include "media_file_defines.h"
+#include "media_file.h"
+#include "module_common_types.h"
+#include "resampler.h"
+#include "thread_wrapper.h"
+#include "tick_util.h"
+#include "typedefs.h"
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+    #include "frame_scaler.h"
+    #include "video_coder.h"
+    #include "video_frames_queue.h"
+#endif
+
+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 FileRecorderImpl : public FileRecorder
+{
+public:
+    FileRecorderImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+    virtual ~FileRecorderImpl();
+
+    // FileRecorder functions.
+    virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
+    virtual FileFormats RecordingFileFormat() const;
+    virtual WebRtc_Word32 StartRecordingAudioFile(
+        const WebRtc_Word8* fileName,
+        const CodecInst& codecInst,
+        WebRtc_UWord32 notificationTimeMs,
+        ACMAMRPackingFormat amrFormat = AMRFileStorage);
+    virtual WebRtc_Word32 StartRecordingAudioFile(
+        OutStream& destStream,
+        const CodecInst& codecInst,
+        WebRtc_UWord32 notificationTimeMs,
+        ACMAMRPackingFormat amrFormat = AMRFileStorage);
+    virtual WebRtc_Word32 StopRecording();
+    virtual bool IsRecording() const;
+    virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const;
+    virtual WebRtc_Word32 RecordAudioToFile(
+        const AudioFrame& frame,
+        const TickTime* playoutTS = NULL);
+    virtual WebRtc_Word32 StartRecordingVideoFile(
+        const WebRtc_Word8* fileName,
+        const CodecInst& audioCodecInst,
+        const VideoCodec& videoCodecInst,
+        ACMAMRPackingFormat amrFormat = AMRFileStorage,
+        bool videoOnly = false)
+    {
+        return -1;
+    }
+    virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame)
+    {
+        return -1;
+    }
+
+protected:
+    virtual WebRtc_Word32 WriteEncodedAudioData(
+        const WebRtc_Word8* audioBuffer,
+        WebRtc_UWord16 bufferLength,
+        WebRtc_UWord16 millisecondsOfData,
+        const TickTime* playoutTS);
+
+    WebRtc_Word32 SetUpAudioEncoder();
+
+    WebRtc_UWord32 _instanceID;
+    FileFormats _fileFormat;
+    MediaFile* _moduleFile;
+
+private:
+    OutStream* _stream;
+    CodecInst codec_info_;
+    ACMAMRPackingFormat _amrFormat;
+
+    WebRtc_Word8 _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
+    AudioCoder _audioEncoder;
+    Resampler _audioResampler;
+};
+
+
+#ifdef WEBRTC_MODULE_UTILITY_VIDEO
+class AviRecorder : public FileRecorderImpl
+{
+public:
+    AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat);
+    virtual ~AviRecorder();
+
+    // FileRecorder functions.
+    virtual WebRtc_Word32 StartRecordingVideoFile(
+        const WebRtc_Word8* fileName,
+        const CodecInst& audioCodecInst,
+        const VideoCodec& videoCodecInst,
+        ACMAMRPackingFormat amrFormat = AMRFileStorage,
+        bool videoOnly = false);
+    virtual WebRtc_Word32 StopRecording();
+    virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame);
+
+protected:
+    virtual WebRtc_Word32 WriteEncodedAudioData(
+        const WebRtc_Word8*  audioBuffer,
+        WebRtc_UWord16 bufferLength,
+        WebRtc_UWord16 millisecondsOfData,
+        const TickTime* playoutTS);
+private:
+    static bool Run(ThreadObj threadObj);
+    bool Process();
+
+    bool StartThread();
+    bool StopThread();
+
+    WebRtc_Word32 EncodeAndWriteVideoToFile(VideoFrame& videoFrame);
+    WebRtc_Word32 ProcessAudio();
+
+    WebRtc_Word32 CalcI420FrameSize() const;
+    WebRtc_Word32 SetUpVideoEncoder();
+
+    VideoCodec _videoCodecInst;
+    bool _videoOnly;
+
+    ListWrapper _audioFramesToWrite;
+    bool _firstAudioFrameReceived;
+
+    VideoFramesQueue* _videoFramesQueue;
+
+    FrameScaler* _frameScaler;
+    VideoCoder* _videoEncoder;
+    WebRtc_Word32 _videoMaxPayloadSize;
+    EncodedVideoData _videoEncodedData;
+
+    ThreadWrapper* _thread;
+    EventWrapper& _timeEvent;
+    CriticalSectionWrapper& _critSec;
+    WebRtc_Word64 _writtenVideoFramesCounter;
+    WebRtc_Word64 _writtenAudioMS;
+    WebRtc_Word64 _writtenVideoMS;
+};
+#endif // WEBRTC_MODULE_UTILITY_VIDEO
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_