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 13 matching lines...) Expand all Loading... | |
24 | 24 |
25 namespace webrtc { | 25 namespace webrtc { |
26 | 26 |
27 // WORK IN PROGRESS | 27 // WORK IN PROGRESS |
28 // This class is under development and is not yet intended for for use outside | 28 // This class is under development and is not yet intended for for use outside |
29 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | 29 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. |
30 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | 30 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 |
31 | 31 |
32 class AudioSendStream { | 32 class AudioSendStream { |
33 public: | 33 public: |
34 virtual ~AudioSendStream() = default; | |
danilchap
2017/07/25 14:42:20
nit: put destructor after child types.
eladalon
2017/07/26 08:43:05
Done.
| |
35 | |
34 struct Stats { | 36 struct Stats { |
35 Stats(); | 37 Stats(); |
36 ~Stats(); | 38 ~Stats(); |
37 | 39 |
38 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. | 40 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. |
39 uint32_t local_ssrc = 0; | 41 uint32_t local_ssrc = 0; |
40 int64_t bytes_sent = 0; | 42 int64_t bytes_sent = 0; |
41 int32_t packets_sent = 0; | 43 int32_t packets_sent = 0; |
42 int32_t packets_lost = -1; | 44 int32_t packets_lost = -1; |
43 float fraction_lost = -1.0f; | 45 float fraction_lost = -1.0f; |
(...skipping 96 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
140 virtual void Stop() = 0; | 142 virtual void Stop() = 0; |
141 | 143 |
142 // TODO(solenberg): Make payload_type a config property instead. | 144 // TODO(solenberg): Make payload_type a config property instead. |
143 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, | 145 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, |
144 int event, int duration_ms) = 0; | 146 int event, int duration_ms) = 0; |
145 | 147 |
146 virtual void SetMuted(bool muted) = 0; | 148 virtual void SetMuted(bool muted) = 0; |
147 | 149 |
148 virtual Stats GetStats() const = 0; | 150 virtual Stats GetStats() const = 0; |
149 | 151 |
150 protected: | 152 virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0; |
the sun
2017/07/25 21:26:49
nit: try order methods the same in interface.h, im
eladalon
2017/07/26 08:43:05
Done.
| |
151 virtual ~AudioSendStream() {} | |
152 }; | 153 }; |
153 } // namespace webrtc | 154 } // namespace webrtc |
154 | 155 |
155 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ | 156 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ |
OLD | NEW |