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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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54 | 54 |
55 // Receive-stream specific RTP settings. | 55 // Receive-stream specific RTP settings. |
56 struct Rtp { | 56 struct Rtp { |
57 std::string ToString() const; | 57 std::string ToString() const; |
58 | 58 |
59 // Sender SSRC. | 59 // Sender SSRC. |
60 uint32_t ssrc = 0; | 60 uint32_t ssrc = 0; |
61 | 61 |
62 // RTP header extensions used for the received stream. | 62 // RTP header extensions used for the received stream. |
63 std::vector<RtpExtension> extensions; | 63 std::vector<RtpExtension> extensions; |
| 64 |
| 65 // RTCP CNAME, see RFC 3550. |
| 66 std::string c_name; |
64 } rtp; | 67 } rtp; |
65 | 68 |
66 // Transport for outgoing packets. The transport is expected to exist for | 69 // Transport for outgoing packets. The transport is expected to exist for |
67 // the entire life of the AudioSendStream and is owned by the API client. | 70 // the entire life of the AudioSendStream and is owned by the API client. |
68 Transport* send_transport = nullptr; | 71 Transport* send_transport = nullptr; |
69 | 72 |
70 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level | 73 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level |
71 // components. | 74 // components. |
72 // TODO(solenberg): Remove when VoiceEngine channels are created outside | 75 // TODO(solenberg): Remove when VoiceEngine channels are created outside |
73 // of Call. | 76 // of Call. |
74 int voe_channel_id = -1; | 77 int voe_channel_id = -1; |
75 | 78 |
76 // Ownership of the encoder object is transferred to Call when the config is | 79 // Ownership of the encoder object is transferred to Call when the config is |
77 // passed to Call::CreateAudioSendStream(). | 80 // passed to Call::CreateAudioSendStream(). |
78 // TODO(solenberg): Implement, once we configure codecs through the new API. | 81 // TODO(solenberg): Implement, once we configure codecs through the new API. |
79 // rtc::scoped_ptr<AudioEncoder> encoder; | 82 // rtc::scoped_ptr<AudioEncoder> encoder; |
80 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. | 83 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. |
81 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. | 84 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. |
82 }; | 85 }; |
83 | 86 |
84 virtual Stats GetStats() const = 0; | 87 virtual Stats GetStats() const = 0; |
85 }; | 88 }; |
86 } // namespace webrtc | 89 } // namespace webrtc |
87 | 90 |
88 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ | 91 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ |
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