Files
platform-external-webrtc/webrtc/modules/utility/source/file_player_impl.h
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

80 lines
2.6 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/media_file/interface/media_file.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/modules/utility/interface/file_player.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class FilePlayerImpl : public FilePlayer
{
public:
FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
~FilePlayerImpl();
virtual int Get10msAudioFromFile(
int16_t* outBuffer,
size_t& lengthInSamples,
int frequencyInHz);
virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
virtual int32_t StartPlayingFile(
const char* fileName,
bool loop,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL);
virtual int32_t StartPlayingFile(
InStream& sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL);
virtual int32_t StopPlayingFile();
virtual bool IsPlayingFile() const;
virtual int32_t GetPlayoutPosition(uint32_t& durationMs);
virtual int32_t AudioCodec(CodecInst& audioCodec) const;
virtual int32_t Frequency() const;
virtual int32_t SetAudioScaling(float scaleFactor);
protected:
int32_t SetUpAudioDecoder();
uint32_t _instanceID;
const FileFormats _fileFormat;
MediaFile& _fileModule;
uint32_t _decodedLengthInMS;
private:
AudioCoder _audioDecoder;
CodecInst _codec;
int32_t _numberOf10MsPerFrame;
int32_t _numberOf10MsInDecoder;
Resampler _resampler;
float _scaling;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_