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Side by Side Diff: webrtc/audio_send_stream.h

Issue 1397123003: Add AudioSendStream to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: more tests Created 5 years, 2 months ago
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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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39 std::string ToString() const; 39 std::string ToString() const;
40 40
41 // Sender SSRC. 41 // Sender SSRC.
42 uint32_t ssrc = 0; 42 uint32_t ssrc = 0;
43 43
44 // RTP header extensions used for the received stream. 44 // RTP header extensions used for the received stream.
45 std::vector<RtpExtension> extensions; 45 std::vector<RtpExtension> extensions;
46 } rtp; 46 } rtp;
47 47
48 // Transport for outgoing packets. 48 // Transport for outgoing packets.
49 Transport* send_transport = nullptr; 49 Transport* send_transport = nullptr;
tommi 2015/10/14 13:21:18 if you could also explain ownership rules for this
the sun 2015/10/14 14:25:24 Done.
50 50
51 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level 51 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
52 // components. 52 // components.
53 // TODO(solenberg): Remove when VoiceEngine channels are created outside 53 // TODO(solenberg): Remove when VoiceEngine channels are created outside
54 // of Call. 54 // of Call.
55 int voe_channel_id = -1; 55 int voe_channel_id = -1;
56 56
57 rtc::scoped_ptr<AudioEncoder> encoder; 57 // Ownership of the encoder object is transferred to Call when the config is
58 // passed to Call::CreateAudioSendStream().
59 AudioEncoder* encoder = nullptr;
tommi 2015/10/14 13:21:18 I'm having a hard time understanding how the comme
the sun 2015/10/14 14:25:24 The config is *copied* into webrtc::internal::Audi
tommi 2015/10/14 14:49:28 I see... is it copied into the internal one then?
the sun 2015/10/15 12:56:06 Not clearcut what is the best approach here. I dec
58 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. 60 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
59 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. 61 int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
60 }; 62 };
61 63
62 virtual Stats GetStats() const = 0; 64 virtual Stats GetStats() const = 0;
63 }; 65 };
64 } // namespace webrtc 66 } // namespace webrtc
65 67
66 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ 68 #endif // WEBRTC_AUDIO_SEND_STREAM_H_
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