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Issue 1351693005: Read the number of TLs for VP9 too + cleanup (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase master Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
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105 105
106 std::vector<VideoStream> streams; 106 std::vector<VideoStream> streams;
107 ContentType content_type; 107 ContentType content_type;
108 void* encoder_specific_settings; 108 void* encoder_specific_settings;
109 109
110 // Padding will be used up to this bitrate regardless of the bitrate produced 110 // Padding will be used up to this bitrate regardless of the bitrate produced
111 // by the encoder. Padding above what's actually produced by the encoder helps 111 // by the encoder. Padding above what's actually produced by the encoder helps
112 // maintaining a higher bitrate estimate. Padding will however not be sent 112 // maintaining a higher bitrate estimate. Padding will however not be sent
113 // unless the estimated bandwidth indicates that the link can handle it. 113 // unless the estimated bandwidth indicates that the link can handle it.
114 int min_transmit_bitrate_bps; 114 int min_transmit_bitrate_bps;
115
116 // disables the generic frame dropper which job it is to drop frames if the
mflodman 2015/10/06 10:49:14 s/disables/Disables
117 // encoder produces a bitrate which is above the target bitrate.
118 bool disable_generic_bitrate_frame_dropper;
115 }; 119 };
116 120
117 // Controls the capacity of the packet buffer in NetEq. The capacity is the 121 // Controls the capacity of the packet buffer in NetEq. The capacity is the
118 // maximum number of packets that the buffer can contain. If the limit is 122 // maximum number of packets that the buffer can contain. If the limit is
119 // exceeded, the buffer will be flushed. The capacity does not affect the actual 123 // exceeded, the buffer will be flushed. The capacity does not affect the actual
120 // audio delay in the general case, since this is governed by the target buffer 124 // audio delay in the general case, since this is governed by the target buffer
121 // level (calculated from the jitter profile). It is only in the rare case of 125 // level (calculated from the jitter profile). It is only in the rare case of
122 // severe network freezes that a higher capacity will lead to a (transient) 126 // severe network freezes that a higher capacity will lead to a (transient)
123 // increase in audio delay. 127 // increase in audio delay.
124 struct NetEqCapacityConfig { 128 struct NetEqCapacityConfig {
125 NetEqCapacityConfig() : enabled(false), capacity(0) {} 129 NetEqCapacityConfig() : enabled(false), capacity(0) {}
126 explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {} 130 explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {}
127 bool enabled; 131 bool enabled;
128 int capacity; 132 int capacity;
129 }; 133 };
130 134
131 struct NetEqFastAccelerate { 135 struct NetEqFastAccelerate {
132 NetEqFastAccelerate() : enabled(false) {} 136 NetEqFastAccelerate() : enabled(false) {}
133 explicit NetEqFastAccelerate(bool value) : enabled(value) {} 137 explicit NetEqFastAccelerate(bool value) : enabled(value) {}
134 bool enabled; 138 bool enabled;
135 }; 139 };
136 140
137 } // namespace webrtc 141 } // namespace webrtc
138 142
139 #endif // WEBRTC_CONFIG_H_ 143 #endif // WEBRTC_CONFIG_H_
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