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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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83 | 83 |
84 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level | 84 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level |
85 // components. | 85 // components. |
86 // TODO(solenberg): Remove when VoiceEngine channels are created outside | 86 // TODO(solenberg): Remove when VoiceEngine channels are created outside |
87 // of Call. | 87 // of Call. |
88 int voe_channel_id = -1; | 88 int voe_channel_id = -1; |
89 | 89 |
90 // Bitrate limits used for variable audio bitrate streams. Set both to -1 to | 90 // Bitrate limits used for variable audio bitrate streams. Set both to -1 to |
91 // disable audio bitrate adaptation. | 91 // disable audio bitrate adaptation. |
92 // Note: This is still an experimental feature and not ready for real usage. | 92 // Note: This is still an experimental feature and not ready for real usage. |
93 int min_bitrate_kbps = -1; | 93 int min_bitrate_bps = -1; |
94 int max_bitrate_kbps = -1; | 94 int max_bitrate_bps = -1; |
95 | 95 |
96 // Defines whether to turn on audio network adaptor, and defines its config | 96 // Defines whether to turn on audio network adaptor, and defines its config |
97 // string. | 97 // string. |
98 rtc::Optional<std::string> audio_network_adaptor_config; | 98 rtc::Optional<std::string> audio_network_adaptor_config; |
99 | 99 |
100 struct SendCodecSpec { | 100 struct SendCodecSpec { |
101 SendCodecSpec(); | 101 SendCodecSpec(); |
102 std::string ToString() const; | 102 std::string ToString() const; |
103 | 103 |
104 bool operator==(const SendCodecSpec& rhs) const; | 104 bool operator==(const SendCodecSpec& rhs) const; |
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133 virtual void SetMuted(bool muted) = 0; | 133 virtual void SetMuted(bool muted) = 0; |
134 | 134 |
135 virtual Stats GetStats() const = 0; | 135 virtual Stats GetStats() const = 0; |
136 | 136 |
137 protected: | 137 protected: |
138 virtual ~AudioSendStream() {} | 138 virtual ~AudioSendStream() {} |
139 }; | 139 }; |
140 } // namespace webrtc | 140 } // namespace webrtc |
141 | 141 |
142 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | 142 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ |
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