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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 |
| (...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; |
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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.
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| 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_ |
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