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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 |
11 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 11 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
12 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 12 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/audio_codecs/audio_encoder_factory.h" | 18 #include "webrtc/api/audio_codecs/audio_encoder_factory.h" |
19 #include "webrtc/api/audio_codecs/audio_format.h" | 19 #include "webrtc/api/audio_codecs/audio_format.h" |
20 #include "webrtc/api/call/transport.h" | 20 #include "webrtc/api/call/transport.h" |
| 21 #include "webrtc/common_types.h" |
21 #include "webrtc/config.h" | 22 #include "webrtc/config.h" |
22 #include "webrtc/rtc_base/optional.h" | 23 #include "webrtc/rtc_base/optional.h" |
23 #include "webrtc/typedefs.h" | 24 #include "webrtc/typedefs.h" |
24 | 25 |
25 namespace webrtc { | 26 namespace webrtc { |
26 | 27 |
27 // WORK IN PROGRESS | 28 // WORK IN PROGRESS |
28 // This class is under development and is not yet intended for for use outside | 29 // This class is under development and is not yet intended for for use outside |
29 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 30 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
30 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 31 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
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52 double total_input_energy = 0.0; | 53 double total_input_energy = 0.0; |
53 double total_input_duration = 0.0; | 54 double total_input_duration = 0.0; |
54 float aec_quality_min = -1.0f; | 55 float aec_quality_min = -1.0f; |
55 int32_t echo_delay_median_ms = -1; | 56 int32_t echo_delay_median_ms = -1; |
56 int32_t echo_delay_std_ms = -1; | 57 int32_t echo_delay_std_ms = -1; |
57 int32_t echo_return_loss = -100; | 58 int32_t echo_return_loss = -100; |
58 int32_t echo_return_loss_enhancement = -100; | 59 int32_t echo_return_loss_enhancement = -100; |
59 float residual_echo_likelihood = -1.0f; | 60 float residual_echo_likelihood = -1.0f; |
60 float residual_echo_likelihood_recent_max = -1.0f; | 61 float residual_echo_likelihood_recent_max = -1.0f; |
61 bool typing_noise_detected = false; | 62 bool typing_noise_detected = false; |
| 63 ANAStats ana_statistics; |
62 }; | 64 }; |
63 | 65 |
64 struct Config { | 66 struct Config { |
65 Config() = delete; | 67 Config() = delete; |
66 explicit Config(Transport* send_transport); | 68 explicit Config(Transport* send_transport); |
67 ~Config(); | 69 ~Config(); |
68 std::string ToString() const; | 70 std::string ToString() const; |
69 | 71 |
70 // Send-stream specific RTP settings. | 72 // Send-stream specific RTP settings. |
71 struct Rtp { | 73 struct Rtp { |
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147 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, | 149 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, |
148 int event, int duration_ms) = 0; | 150 int event, int duration_ms) = 0; |
149 | 151 |
150 virtual void SetMuted(bool muted) = 0; | 152 virtual void SetMuted(bool muted) = 0; |
151 | 153 |
152 virtual Stats GetStats() const = 0; | 154 virtual Stats GetStats() const = 0; |
153 }; | 155 }; |
154 } // namespace webrtc | 156 } // namespace webrtc |
155 | 157 |
156 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 158 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
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