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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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45 int32_t ext_seqnum = -1; | 45 int32_t ext_seqnum = -1; |
46 int32_t jitter_ms = -1; | 46 int32_t jitter_ms = -1; |
47 int64_t rtt_ms = -1; | 47 int64_t rtt_ms = -1; |
48 int32_t audio_level = -1; | 48 int32_t audio_level = -1; |
49 float aec_quality_min = -1.0f; | 49 float aec_quality_min = -1.0f; |
50 int32_t echo_delay_median_ms = -1; | 50 int32_t echo_delay_median_ms = -1; |
51 int32_t echo_delay_std_ms = -1; | 51 int32_t echo_delay_std_ms = -1; |
52 int32_t echo_return_loss = -100; | 52 int32_t echo_return_loss = -100; |
53 int32_t echo_return_loss_enhancement = -100; | 53 int32_t echo_return_loss_enhancement = -100; |
54 float residual_echo_likelihood = -1.0f; | 54 float residual_echo_likelihood = -1.0f; |
| 55 float residual_echo_likelihood_recent_max = -1.0f; |
55 bool typing_noise_detected = false; | 56 bool typing_noise_detected = false; |
56 }; | 57 }; |
57 | 58 |
58 struct Config { | 59 struct Config { |
59 Config() = delete; | 60 Config() = delete; |
60 explicit Config(Transport* send_transport); | 61 explicit Config(Transport* send_transport); |
61 ~Config(); | 62 ~Config(); |
62 std::string ToString() const; | 63 std::string ToString() const; |
63 | 64 |
64 // Send-stream specific RTP settings. | 65 // Send-stream specific RTP settings. |
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136 virtual void SetMuted(bool muted) = 0; | 137 virtual void SetMuted(bool muted) = 0; |
137 | 138 |
138 virtual Stats GetStats() const = 0; | 139 virtual Stats GetStats() const = 0; |
139 | 140 |
140 protected: | 141 protected: |
141 virtual ~AudioSendStream() {} | 142 virtual ~AudioSendStream() {} |
142 }; | 143 }; |
143 } // namespace webrtc | 144 } // namespace webrtc |
144 | 145 |
145 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 146 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
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