Remove include dirs from modules/{media_file,pacing}

webrtc/modules/media_file/source/media_file_utility.h

Index: webrtc/modules/media_file/source/media_file_utility.h
diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/source/media_file_utility.h
deleted file mode 100644
index 65921bd9638b3380328077126a3357b7edd3cca8..0000000000000000000000000000000000000000
--- a/webrtc/modules/media_file/source/media_file_utility.h
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@@ -1,284 +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.
- */
-
-// Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-
-#include <stdio.h>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/include/media_file_defines.h"
-
-namespace webrtc {
-class InStream;
-class OutStream;
-
-class ModuleFileUtility
-{
-public:
-
-    ModuleFileUtility(const int32_t id);
-    ~ModuleFileUtility();
-
-    // Prepare for playing audio from stream.
-    // startPointMs and stopPointMs, unless zero, specify what part of the file
-    // should be read. From startPointMs ms to stopPointMs ms.
-    int32_t InitWavReading(InStream& stream,
-                           const uint32_t startPointMs = 0,
-                           const uint32_t stopPointMs = 0);
-
-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
-    // The return value is the number of bytes written to audioBuffer.
-    // Note: This API only play mono audio but can be used on file containing
-    // audio with more channels (in which case the audio will be converted to
-    // mono).
-    int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
-                              const size_t dataLengthInBytes);
-
-    // Put 10-60ms, depending on codec frame size, of audio data from file into
-    // audioBufferLeft and audioBufferRight. The buffers contain the left and
-    // right channel of played out stereo audio.
-    // dataLengthInBytes  indicates the size of both audioBufferLeft and
-    // audioBufferRight.
-    // The return value is the number of bytes read for each buffer.
-    // Note: This API can only be successfully called for WAV files with stereo
-    // audio.
-    int32_t ReadWavDataAsStereo(InStream& wav,
-                                int8_t* audioBufferLeft,
-                                int8_t* audioBufferRight,
-                                const size_t bufferLength);
-
-    // Prepare for recording audio to stream.
-    // codecInst specifies the encoding of the audio data.
-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
-    // mono). Stereo is only supported for WAV files.
-    int32_t InitWavWriting(OutStream& stream, const CodecInst& codecInst);
-
-    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
-    // to file. The audio frame size is determined by the codecInst.pacsize
-    // parameter of the last sucessfull StartRecordingAudioFile(..) call.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t WriteWavData(OutStream& stream,
-                         const int8_t* audioBuffer,
-                         const size_t bufferLength);
-
-    // Finalizes the WAV header so that it is correct if nothing more will be
-    // written to stream.
-    // Note: this API must be called before closing stream to ensure that the
-    //       WAVE header is updated with the file size. Don't call this API
-    //       if more samples are to be written to stream.
-    int32_t UpdateWavHeader(OutStream& stream);
-
-    // Prepare for playing audio from stream.
-    // startPointMs and stopPointMs, unless zero, specify what part of the file
-    // should be read. From startPointMs ms to stopPointMs ms.
-    // freqInHz is the PCM sampling frequency.
-    // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
-    int32_t InitPCMReading(InStream& stream,
-                           const uint32_t startPointMs = 0,
-                           const uint32_t stopPointMs = 0,
-                           const uint32_t freqInHz = 16000);
-
-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t ReadPCMData(InStream& stream, int8_t* audioBuffer,
-                        const size_t dataLengthInBytes);
-
-    // Prepare for recording audio to stream.
-    // freqInHz is the PCM sampling frequency.
-    // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
-    int32_t InitPCMWriting(OutStream& stream, const uint32_t freqInHz = 16000);
-
-    // Write one 10ms audio frame, i.e. the bufferLength first bytes of
-    // audioBuffer, to file. The audio frame size is determined by the freqInHz
-    // parameter of the last sucessfull InitPCMWriting(..) call.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t WritePCMData(OutStream& stream,
-                         const int8_t* audioBuffer,
-                         size_t bufferLength);
-
-    // Prepare for playing audio from stream.
-    // startPointMs and stopPointMs, unless zero, specify what part of the file
-    // should be read. From startPointMs ms to stopPointMs ms.
-    int32_t InitCompressedReading(InStream& stream,
-                                  const uint32_t startPointMs = 0,
-                                  const uint32_t stopPointMs = 0);
-
-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t ReadCompressedData(InStream& stream,
-                               int8_t* audioBuffer,
-                               const size_t dataLengthInBytes);
-
-    // Prepare for recording audio to stream.
-    // codecInst specifies the encoding of the audio data.
-    int32_t InitCompressedWriting(OutStream& stream,
-                                  const CodecInst& codecInst);
-
-    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
-    // to file. The audio frame size is determined by the codecInst.pacsize
-    // parameter of the last sucessfull InitCompressedWriting(..) call.
-    // The return value is the number of bytes written to stream.
-    // Note: bufferLength must be exactly one frame.
-    int32_t WriteCompressedData(OutStream& stream,
-                                const int8_t* audioBuffer,
-                                const size_t bufferLength);
-
-    // Prepare for playing audio from stream.
-    // codecInst specifies the encoding of the audio data.
-    int32_t InitPreEncodedReading(InStream& stream,
-                                  const CodecInst& codecInst);
-
-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t ReadPreEncodedData(InStream& stream,
-                               int8_t* audioBuffer,
-                               const size_t dataLengthInBytes);
-
-    // Prepare for recording audio to stream.
-    // codecInst specifies the encoding of the audio data.
-    int32_t InitPreEncodedWriting(OutStream& stream,
-                                  const CodecInst& codecInst);
-
-    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
-    // to stream. The audio frame size is determined by the codecInst.pacsize
-    // parameter of the last sucessfull InitPreEncodedWriting(..) call.
-   // The return value is the number of bytes written to stream.
-    // Note: bufferLength must be exactly one frame.
-    int32_t WritePreEncodedData(OutStream& stream,
-                                const int8_t* inData,
-                                const size_t dataLengthInBytes);
-
-    // Set durationMs to the size of the file (in ms) specified by fileName.
-    // freqInHz specifies the sampling frequency of the file.
-    int32_t FileDurationMs(const char* fileName,
-                           const FileFormats fileFormat,
-                           const uint32_t freqInHz = 16000);
-
-    // Return the number of ms that have been played so far.
-    uint32_t PlayoutPositionMs();
-
-    // Update codecInst according to the current audio codec being used for
-    // reading or writing.
-    int32_t codec_info(CodecInst& codecInst);
-
-private:
-    // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
-    enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
-
-
-    int32_t InitWavCodec(uint32_t samplesPerSec,
-                         uint32_t channels,
-                         uint32_t bitsPerSample,
-                         uint32_t formatTag);
-
-    // Parse the WAV header in stream.
-    int32_t ReadWavHeader(InStream& stream);
-
-    // Update the WAV header. freqInHz, bytesPerSample, channels, format,
-    // lengthInBytes specify characterists of the audio data.
-    // freqInHz is the sampling frequency. bytesPerSample is the sample size in
-    // bytes. channels is the number of channels, e.g. 1 is mono and 2 is
-    // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
-    // lengthInBytes is the number of bytes the audio samples are using up.
-    int32_t WriteWavHeader(OutStream& stream,
-                           const uint32_t freqInHz,
-                           const uint32_t bytesPerSample,
-                           const uint32_t channels,
-                           const uint32_t format,
-                           const uint32_t lengthInBytes);
-
-    // Put dataLengthInBytes of audio data from stream into the audioBuffer.
-    // The return value is the number of bytes written to audioBuffer.
-    int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
-                        const uint32_t dataLengthInBytes);
-
-    // Update the current audio codec being used for reading or writing
-    // according to codecInst.
-    int32_t set_codec_info(const CodecInst& codecInst);
-
-    struct WAVE_FMTINFO_header
-    {
-        int16_t formatTag;
-        int16_t nChannels;
-        int32_t nSamplesPerSec;
-        int32_t nAvgBytesPerSec;
-        int16_t nBlockAlign;
-        int16_t nBitsPerSample;
-    };
-    // Identifiers for preencoded files.
-    enum MediaFileUtility_CodecType
-    {
-        kCodecNoCodec  = 0,
-        kCodecIsac,
-        kCodecIsacSwb,
-        kCodecIsacLc,
-        kCodecL16_8Khz,
-        kCodecL16_16kHz,
-        kCodecL16_32Khz,
-        kCodecPcmu,
-        kCodecPcma,
-        kCodecIlbc20Ms,
-        kCodecIlbc30Ms,
-        kCodecG722,
-        kCodecG722_1_32Kbps,
-        kCodecG722_1_24Kbps,
-        kCodecG722_1_16Kbps,
-        kCodecG722_1c_48,
-        kCodecG722_1c_32,
-        kCodecG722_1c_24,
-        kCodecAmr,
-        kCodecAmrWb,
-        kCodecG729,
-        kCodecG729_1,
-        kCodecG726_40,
-        kCodecG726_32,
-        kCodecG726_24,
-        kCodecG726_16,
-        kCodecSpeex8Khz,
-        kCodecSpeex16Khz
-    };
-
-    // TODO (hellner): why store multiple formats. Just store either codec_info_
-    //                 or _wavFormatObj and supply conversion functions.
-    WAVE_FMTINFO_header _wavFormatObj;
-    int32_t _dataSize;      // Chunk size if reading a WAV file
-    // Number of bytes to read. I.e. frame size in bytes. May be multiple
-    // chunks if reading WAV.
-    int32_t _readSizeBytes;
-
-    int32_t _id;
-
-    uint32_t _stopPointInMs;
-    uint32_t _startPointInMs;
-    uint32_t _playoutPositionMs;
-    size_t _bytesWritten;
-
-    CodecInst codec_info_;
-    MediaFileUtility_CodecType _codecId;
-
-    // The amount of bytes, on average, used for one audio sample.
-    int32_t  _bytesPerSample;
-    int32_t  _readPos;
-
-    // Only reading or writing can be enabled, not both.
-    bool _reading;
-    bool _writing;
-
-    // Scratch buffer used for turning stereo audio to mono.
-    uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_