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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 11 #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
12 #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 12 #define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
13 | 13 |
14 #include "webrtc/common_types.h" | 14 #include "webrtc/common_types.h" |
15 #include "webrtc/engine_configurations.h" | 15 #include "webrtc/engine_configurations.h" |
16 #include "webrtc/modules/include/module_common_types.h" | 16 #include "webrtc/modules/include/module_common_types.h" |
17 #include "webrtc/typedefs.h" | 17 #include "webrtc/typedefs.h" |
18 #include "webrtc/video_frame.h" | |
19 | 18 |
20 namespace webrtc { | 19 namespace webrtc { |
21 class FileCallback; | 20 class FileCallback; |
22 | 21 |
23 class FilePlayer | 22 class FilePlayer |
24 { | 23 { |
25 public: | 24 public: |
26 // The largest decoded frame size in samples (60ms with 32kHz sample rate). | 25 // The largest decoded frame size in samples (60ms with 32kHz sample rate). |
27 enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32}; | 26 enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32}; |
28 enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2}; | 27 enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2}; |
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72 virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0; | 71 virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0; |
73 | 72 |
74 // Set audioCodec to the currently used audio codec. | 73 // Set audioCodec to the currently used audio codec. |
75 virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0; | 74 virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0; |
76 | 75 |
77 virtual int32_t Frequency() const = 0; | 76 virtual int32_t Frequency() const = 0; |
78 | 77 |
79 // Note: scaleFactor is in the range [0.0 - 2.0] | 78 // Note: scaleFactor is in the range [0.0 - 2.0] |
80 virtual int32_t SetAudioScaling(float scaleFactor) = 0; | 79 virtual int32_t SetAudioScaling(float scaleFactor) = 0; |
81 | 80 |
82 // Return the time in ms until next video frame should be pulled (by | |
83 // calling GetVideoFromFile(..)). | |
84 // Note: this API reads one video frame from file. This means that it should | |
85 // be called exactly once per GetVideoFromFile(..) API call. | |
86 virtual int32_t TimeUntilNextVideoFrame() { return -1;} | |
87 | |
88 virtual int32_t StartPlayingVideoFile( | |
89 const char* /*fileName*/, | |
90 bool /*loop*/, | |
91 bool /*videoOnly*/) { return -1;} | |
92 | |
93 virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const | |
94 {return -1;} | |
95 | |
96 virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; } | |
97 | |
98 // Same as GetVideoFromFile(). videoFrame will have the resolution specified | |
99 // by the width outWidth and height outHeight in pixels. | |
100 virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/, | |
101 const uint32_t /*outWidth*/, | |
102 const uint32_t /*outHeight*/) { | |
103 return -1; | |
104 } | |
105 | |
106 protected: | 81 protected: |
107 virtual ~FilePlayer() {} | 82 virtual ~FilePlayer() {} |
108 | 83 |
109 }; | 84 }; |
110 } // namespace webrtc | 85 } // namespace webrtc |
111 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ | 86 #endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ |
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