Line data Source code
1 : /*
2 : * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 : *
4 : * Use of this source code is governed by a BSD-style license
5 : * that can be found in the LICENSE file in the root of the source
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : */
10 :
11 : #ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
12 : #define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
13 :
14 : #include <memory>
15 :
16 : #include "webrtc/common_types.h"
17 : #include "webrtc/modules/include/module_common_types.h"
18 : #include "webrtc/typedefs.h"
19 :
20 : namespace webrtc {
21 :
22 : class FileCallback;
23 :
24 0 : class FilePlayer {
25 : public:
26 : // The largest decoded frame size in samples (60ms with 32kHz sample rate).
27 : enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
28 : enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
29 :
30 : // Note: will return NULL for unsupported formats.
31 : static std::unique_ptr<FilePlayer> CreateFilePlayer(
32 : const uint32_t instanceID,
33 : const FileFormats fileFormat);
34 :
35 0 : virtual ~FilePlayer() = default;
36 :
37 : // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
38 : // will be set to the number of samples read (not the number of samples per
39 : // channel).
40 : virtual int Get10msAudioFromFile(int16_t* outBuffer,
41 : size_t* lengthInSamples,
42 : int frequencyInHz) = 0;
43 :
44 : // Register callback for receiving file playing notifications.
45 : virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
46 :
47 : // API for playing audio from fileName to channel.
48 : // Note: codecInst is used for pre-encoded files.
49 : virtual int32_t StartPlayingFile(const char* fileName,
50 : bool loop,
51 : uint32_t startPosition,
52 : float volumeScaling,
53 : uint32_t notification,
54 : uint32_t stopPosition,
55 : const CodecInst* codecInst) = 0;
56 :
57 : // Note: codecInst is used for pre-encoded files.
58 : virtual int32_t StartPlayingFile(InStream* sourceStream,
59 : uint32_t startPosition,
60 : float volumeScaling,
61 : uint32_t notification,
62 : uint32_t stopPosition,
63 : const CodecInst* codecInst) = 0;
64 :
65 : virtual int32_t StopPlayingFile() = 0;
66 :
67 : virtual bool IsPlayingFile() const = 0;
68 :
69 : virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
70 :
71 : // Set audioCodec to the currently used audio codec.
72 : virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
73 :
74 : virtual int32_t Frequency() const = 0;
75 :
76 : // Note: scaleFactor is in the range [0.0 - 2.0]
77 : virtual int32_t SetAudioScaling(float scaleFactor) = 0;
78 : };
79 : } // namespace webrtc
80 : #endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
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