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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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52 double total_input_energy = 0.0; | 52 double total_input_energy = 0.0; |
53 double total_input_duration = 0.0; | 53 double total_input_duration = 0.0; |
54 float aec_quality_min = -1.0f; | 54 float aec_quality_min = -1.0f; |
55 int32_t echo_delay_median_ms = -1; | 55 int32_t echo_delay_median_ms = -1; |
56 int32_t echo_delay_std_ms = -1; | 56 int32_t echo_delay_std_ms = -1; |
57 int32_t echo_return_loss = -100; | 57 int32_t echo_return_loss = -100; |
58 int32_t echo_return_loss_enhancement = -100; | 58 int32_t echo_return_loss_enhancement = -100; |
59 float residual_echo_likelihood = -1.0f; | 59 float residual_echo_likelihood = -1.0f; |
60 float residual_echo_likelihood_recent_max = -1.0f; | 60 float residual_echo_likelihood_recent_max = -1.0f; |
61 bool typing_noise_detected = false; | 61 bool typing_noise_detected = false; |
| 62 rtc::Optional<int> ana_bitrate_action_counter; |
| 63 rtc::Optional<int> ana_channel_action_counter; |
| 64 rtc::Optional<int> ana_dtx_action_counter; |
| 65 rtc::Optional<int> ana_fec_action_counter; |
| 66 rtc::Optional<int> ana_frame_length_action_counter; |
62 }; | 67 }; |
63 | 68 |
64 struct Config { | 69 struct Config { |
65 Config() = delete; | 70 Config() = delete; |
66 explicit Config(Transport* send_transport); | 71 explicit Config(Transport* send_transport); |
67 ~Config(); | 72 ~Config(); |
68 std::string ToString() const; | 73 std::string ToString() const; |
69 | 74 |
70 // Send-stream specific RTP settings. | 75 // Send-stream specific RTP settings. |
71 struct Rtp { | 76 struct Rtp { |
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147 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, | 152 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, |
148 int event, int duration_ms) = 0; | 153 int event, int duration_ms) = 0; |
149 | 154 |
150 virtual void SetMuted(bool muted) = 0; | 155 virtual void SetMuted(bool muted) = 0; |
151 | 156 |
152 virtual Stats GetStats() const = 0; | 157 virtual Stats GetStats() const = 0; |
153 }; | 158 }; |
154 } // namespace webrtc | 159 } // namespace webrtc |
155 | 160 |
156 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 161 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
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