bc436ede07
Reason for revert:
broke internal project
Original issue's description:
> Supporting 48kHz PCM file.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2790493004
> Cr-Commit-Position: refs/heads/master@{#17493}
> Committed: 5f93709e7c
TBR=niklas.enbom@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2791453004
Cr-Commit-Position: refs/heads/master@{#17496}
81 lines
3.0 KiB
C++
81 lines
3.0 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_VOICE_ENGINE_FILE_PLAYER_H_
|
|
#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
|
|
|
|
#include <memory>
|
|
|
|
#include "webrtc/common_types.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> CreateFilePlayer(
|
|
const uint32_t instanceID,
|
|
const FileFormats fileFormat);
|
|
|
|
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;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
|