237 lines
11 KiB
C++
237 lines
11 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
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#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
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#include "common_types.h"
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#include "typedefs.h"
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#include "module.h"
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#include "module_common_types.h"
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#include "media_file_defines.h"
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namespace webrtc {
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class MediaFile : public Module
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{
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public:
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// Factory method. Constructor disabled. id is the identifier for the
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// MediaFile instance.
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static MediaFile* CreateMediaFile(const WebRtc_Word32 id);
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static void DestroyMediaFile(MediaFile* module);
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// Writes the version of the MediaFile to version. remainingBufferInBytes
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// is both an input parameter and an output parameter. It indicates the size
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// of version less messages in it. position is both an input parameter and
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// an output parameter. It indicates the position of the NULL termination
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// in the version string.
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static WebRtc_Word32 GetVersion(WebRtc_Word8* version,
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WebRtc_UWord32& remainingBufferInBytes,
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WebRtc_UWord32& position);
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// Set the MediaFile instance identifier.
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
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// Put 10-60ms of audio data from file into the audioBuffer depending on
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// codec frame size. dataLengthInBytes is both an input and output
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// parameter. As input parameter it indicates the size of audioBuffer.
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// As output parameter it indicates the number of bytes written to
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// audioBuffer.
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// Note: This API only play mono audio but can be used on file containing
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// audio with more channels (in which case the audio will be converted to
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// mono).
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virtual WebRtc_Word32 PlayoutAudioData(
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WebRtc_Word8* audioBuffer,
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WebRtc_UWord32& dataLengthInBytes) = 0;
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// Put one video frame into videoBuffer. dataLengthInBytes is both an input
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// and output parameter. As input parameter it indicates the size of
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// videoBuffer. As output parameter it indicates the number of bytes written
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// to videoBuffer.
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virtual WebRtc_Word32 PlayoutAVIVideoData(
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WebRtc_Word8* videoBuffer,
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WebRtc_UWord32& dataLengthInBytes) = 0;
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// Put 10-60ms, depending on codec frame size, of audio data from file into
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// audioBufferLeft and audioBufferRight. The buffers contain the left and
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// right channel of played out stereo audio.
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// dataLengthInBytes is both an input and output parameter. As input
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// parameter it indicates the size of both audioBufferLeft and
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// audioBufferRight. As output parameter it indicates the number of bytes
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// written to both audio buffers.
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// Note: This API can only be successfully called for WAV files with stereo
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// audio.
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virtual WebRtc_Word32 PlayoutStereoData(
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WebRtc_Word8* audioBufferLeft,
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WebRtc_Word8* audioBufferRight,
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WebRtc_UWord32& dataLengthInBytes) = 0;
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// Open the file specified by fileName (relative path is allowed) for
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// reading. FileCallback::PlayNotification(..) will be called after
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// notificationTimeMs of the file has been played if notificationTimeMs is
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// greater than zero. If loop is true the file will be played until
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// StopPlaying() is called. When end of file is reached the file is read
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// from the start. format specifies the type of file fileName refers to.
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// codecInst specifies the encoding of the audio data. Note that
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// file formats that contain this information (like WAV files) don't need to
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// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
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// specify what part of the file should be read. From startPointMs ms to
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// stopPointMs ms.
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// Note: codecInst.channels should be set to 2 for stereo (and 1 for
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// mono). Stereo audio is only supported for WAV files.
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virtual WebRtc_Word32 StartPlayingAudioFile(
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const WebRtc_Word8* fileName,
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const WebRtc_UWord32 notificationTimeMs = 0,
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const bool loop = false,
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const FileFormats format = kFileFormatPcm16kHzFile,
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const CodecInst* codecInst = NULL,
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const WebRtc_UWord32 startPointMs = 0,
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const WebRtc_UWord32 stopPointMs = 0) = 0;
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// Open the file specified by fileName for reading (relative path is
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// allowed). If loop is true the file will be played until StopPlaying() is
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// called. When end of file is reached the file is read from the start.
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// format specifies the type of file fileName refers to. Only video will be
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// read if videoOnly is true.
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virtual WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
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const bool loop,
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bool videoOnly,
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const FileFormats format) = 0;
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// Prepare for playing audio from stream.
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// FileCallback::PlayNotification(..) will be called after
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// notificationTimeMs of the file has been played if notificationTimeMs is
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// greater than zero. format specifies the type of file fileName refers to.
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// codecInst specifies the encoding of the audio data. Note that
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// file formats that contain this information (like WAV files) don't need to
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// provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
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// specify what part of the file should be read. From startPointMs ms to
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// stopPointMs ms.
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// Note: codecInst.channels should be set to 2 for stereo (and 1 for
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// mono). Stereo audio is only supported for WAV files.
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virtual WebRtc_Word32 StartPlayingAudioStream(
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InStream& stream,
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const WebRtc_UWord32 notificationTimeMs = 0,
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const FileFormats format = kFileFormatPcm16kHzFile,
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const CodecInst* codecInst = NULL,
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const WebRtc_UWord32 startPointMs = 0,
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const WebRtc_UWord32 stopPointMs = 0) = 0;
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// Stop playing from file or stream.
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virtual WebRtc_Word32 StopPlaying() = 0;
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// Return true if playing.
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virtual bool IsPlaying() = 0;
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// Set durationMs to the number of ms that has been played from file.
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virtual WebRtc_Word32 PlayoutPositionMs(
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WebRtc_UWord32& durationMs) const = 0;
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// Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
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// to file. The audio frame size is determined by the codecInst.pacsize
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// parameter of the last sucessfull StartRecordingAudioFile(..) call.
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// Note: bufferLength must be exactly one frame.
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virtual WebRtc_Word32 IncomingAudioData(
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const WebRtc_Word8* audioBuffer,
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const WebRtc_UWord32 bufferLength) = 0;
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// Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
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// to file.
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// Note: videoBuffer can contain encoded data. The codec used must be the
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// same as what was specified by videoCodecInst for the last successfull
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// StartRecordingVideoFile(..) call. The videoBuffer must contain exactly
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// one video frame.
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virtual WebRtc_Word32 IncomingAVIVideoData(
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const WebRtc_Word8* videoBuffer,
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const WebRtc_UWord32 bufferLength) = 0;
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// Open/creates file specified by fileName for writing (relative path is
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// allowed). FileCallback::RecordNotification(..) will be called after
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// notificationTimeMs of audio data has been recorded if
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// notificationTimeMs is greater than zero.
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// format specifies the type of file that should be created/opened.
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// codecInst specifies the encoding of the audio data. maxSizeBytes
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// specifies the number of bytes allowed to be written to file if it is
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// greater than zero.
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// Note: codecInst.channels should be set to 2 for stereo (and 1 for
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// mono). Stereo is only supported for WAV files.
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virtual WebRtc_Word32 StartRecordingAudioFile(
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const WebRtc_Word8* fileName,
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const FileFormats format,
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const CodecInst& codecInst,
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const WebRtc_UWord32 notificationTimeMs = 0,
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const WebRtc_UWord32 maxSizeBytes = 0) = 0;
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// Open/create the file specified by fileName for writing audio/video data
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// (relative path is allowed). format specifies the type of file fileName
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// should be. codecInst specifies the encoding of the audio data.
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// videoCodecInst specifies the encoding of the video data. Only video data
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// will be recorded if videoOnly is true.
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virtual WebRtc_Word32 StartRecordingVideoFile(
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const WebRtc_Word8* fileName,
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const FileFormats format,
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const CodecInst& codecInst,
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const VideoCodec& videoCodecInst,
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bool videoOnly = false) = 0;
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// Prepare for recording audio to stream.
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// FileCallback::RecordNotification(..) will be called after
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// notificationTimeMs of audio data has been recorded if
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// notificationTimeMs is greater than zero.
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// format specifies the type of file that stream should correspond to.
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// codecInst specifies the encoding of the audio data.
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// Note: codecInst.channels should be set to 2 for stereo (and 1 for
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// mono). Stereo is only supported for WAV files.
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virtual WebRtc_Word32 StartRecordingAudioStream(
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OutStream& stream,
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const FileFormats format,
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const CodecInst& codecInst,
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const WebRtc_UWord32 notificationTimeMs = 0) = 0;
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// Stop recording to file or stream.
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virtual WebRtc_Word32 StopRecording() = 0;
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// Return true if recording.
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virtual bool IsRecording() = 0;
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// Set durationMs to the number of ms that has been recorded to file.
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virtual WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs) = 0;
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// Return true if recording or playing is stereo.
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virtual bool IsStereo() = 0;
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// Register callback to receive media file related notifications. Disables
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// callbacks if callback is NULL.
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virtual WebRtc_Word32 SetModuleFileCallback(FileCallback* callback) = 0;
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// Set durationMs to the size of the file (in ms) specified by fileName.
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// format specifies the type of file fileName refers to. freqInHz specifies
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// the sampling frequency of the file.
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virtual WebRtc_Word32 FileDurationMs(
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const WebRtc_Word8* fileName,
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WebRtc_UWord32& durationMs,
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const FileFormats format,
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const WebRtc_UWord32 freqInHz = 16000) = 0;
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// Update codecInst according to the current audio codec being used for
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// reading or writing.
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virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0;
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// Update videoCodecInst according to the current video codec being used for
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// reading or writing.
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virtual WebRtc_Word32 VideoCodecInst(VideoCodec& videoCodecInst) const = 0;
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protected:
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MediaFile() {}
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virtual ~MediaFile() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
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