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Side by Side Diff: webrtc/modules/utility/include/file_player.h

Issue 2033433004: Remove the unused video stuff in FilePlayer and FileRecorder (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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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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