OLD | NEW |
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 |
(...skipping 29 matching lines...) Expand all Loading... |
40 int64_t bytes_sent = 0; | 40 int64_t bytes_sent = 0; |
41 int32_t packets_sent = 0; | 41 int32_t packets_sent = 0; |
42 int32_t packets_lost = -1; | 42 int32_t packets_lost = -1; |
43 float fraction_lost = -1.0f; | 43 float fraction_lost = -1.0f; |
44 std::string codec_name; | 44 std::string codec_name; |
45 rtc::Optional<int> codec_payload_type; | 45 rtc::Optional<int> codec_payload_type; |
46 int32_t ext_seqnum = -1; | 46 int32_t ext_seqnum = -1; |
47 int32_t jitter_ms = -1; | 47 int32_t jitter_ms = -1; |
48 int64_t rtt_ms = -1; | 48 int64_t rtt_ms = -1; |
49 int32_t audio_level = -1; | 49 int32_t audio_level = -1; |
| 50 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| 51 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaud
ioenergy |
| 52 double total_input_energy = 0.0; |
| 53 double total_input_duration = 0.0; |
50 float aec_quality_min = -1.0f; | 54 float aec_quality_min = -1.0f; |
51 int32_t echo_delay_median_ms = -1; | 55 int32_t echo_delay_median_ms = -1; |
52 int32_t echo_delay_std_ms = -1; | 56 int32_t echo_delay_std_ms = -1; |
53 int32_t echo_return_loss = -100; | 57 int32_t echo_return_loss = -100; |
54 int32_t echo_return_loss_enhancement = -100; | 58 int32_t echo_return_loss_enhancement = -100; |
55 float residual_echo_likelihood = -1.0f; | 59 float residual_echo_likelihood = -1.0f; |
56 float residual_echo_likelihood_recent_max = -1.0f; | 60 float residual_echo_likelihood_recent_max = -1.0f; |
57 bool typing_noise_detected = false; | 61 bool typing_noise_detected = false; |
58 }; | 62 }; |
59 | 63 |
(...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
142 virtual void SetMuted(bool muted) = 0; | 146 virtual void SetMuted(bool muted) = 0; |
143 | 147 |
144 virtual Stats GetStats() const = 0; | 148 virtual Stats GetStats() const = 0; |
145 | 149 |
146 protected: | 150 protected: |
147 virtual ~AudioSendStream() {} | 151 virtual ~AudioSendStream() {} |
148 }; | 152 }; |
149 } // namespace webrtc | 153 } // namespace webrtc |
150 | 154 |
151 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 155 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
OLD | NEW |