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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.
*/
#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#include <memory>
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class FileCallback;
class FilePlayer {
public:
// 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 };
// Note: will return NULL for unsupported formats.
static std::unique_ptr<FilePlayer> NewFilePlayer(
const uint32_t instanceID,
const FileFormats fileFormat);
// Deprecated creation/destruction functions. Use NewFilePlayer instead.
static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
const FileFormats fileFormat);
static void DestroyFilePlayer(FilePlayer* player);
virtual ~FilePlayer() = default;
// Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
// will be set to the number of samples read (not the number of samples per
// channel).
virtual int Get10msAudioFromFile(int16_t* outBuffer,
size_t* lengthInSamples,
int frequencyInHz) = 0;
// Register callback for receiving file playing notifications.
virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
// API for playing audio from fileName to channel.
// Note: codecInst is used for pre-encoded files.
virtual int32_t StartPlayingFile(const char* fileName,
bool loop,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition,
const CodecInst* codecInst) = 0;
// Note: codecInst is used for pre-encoded files.
virtual int32_t StartPlayingFile(InStream* sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition,
const CodecInst* codecInst) = 0;
virtual int32_t StopPlayingFile() = 0;
virtual bool IsPlayingFile() const = 0;
virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
// Set audioCodec to the currently used audio codec.
virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
virtual int32_t Frequency() const = 0;
// Note: scaleFactor is in the range [0.0 - 2.0]
virtual int32_t SetAudioScaling(float scaleFactor) = 0;
// Deprecated functions. Use the functions above with the same name instead.
int Get10msAudioFromFile(int16_t* outBuffer,
size_t& lengthInSamples,
int frequencyInHz) {
return Get10msAudioFromFile(outBuffer, &lengthInSamples, frequencyInHz);
}
int32_t StartPlayingFile(InStream& sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition,
const CodecInst* codecInst) {
return StartPlayingFile(&sourceStream, startPosition, volumeScaling,
notification, stopPosition, codecInst);
}
int32_t GetPlayoutPosition(uint32_t& durationMs) {
return GetPlayoutPosition(&durationMs);
}
int32_t AudioCodec(CodecInst& audioCodec) const {
return AudioCodec(&audioCodec);
}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_