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1 /* | |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #include "webrtc/audio/audio_send_stream.h" | |
12 | |
13 #include <string> | |
14 | |
15 #include "webrtc/base/checks.h" | |
16 #include "webrtc/base/logging.h" | |
17 | |
18 namespace webrtc { | |
19 std::string AudioSendStream::Config::Rtp::ToString() const { | |
20 std::stringstream ss; | |
21 ss << "{ssrc: " << ssrc; | |
22 ss << ", extensions: ["; | |
23 for (size_t i = 0; i < extensions.size(); ++i) { | |
24 ss << extensions[i].ToString(); | |
25 if (i != extensions.size() - 1) | |
26 ss << ", "; | |
27 } | |
28 ss << ']'; | |
29 ss << '}'; | |
30 return ss.str(); | |
31 } | |
32 | |
33 std::string AudioSendStream::Config::ToString() const { | |
34 std::stringstream ss; | |
35 ss << "{rtp: " << rtp.ToString(); | |
36 ss << ", voe_channel_id: " << voe_channel_id; | |
37 // TODO(solenberg): Encoder config. | |
38 ss << ", cng_payload_type: " << cng_payload_type; | |
39 ss << ", red_payload_type: " << red_payload_type; | |
40 ss << '}'; | |
41 return ss.str(); | |
42 } | |
43 | |
44 namespace internal { | |
45 AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config) | |
46 : config_(config) { | |
47 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); | |
48 RTC_DCHECK(config.voe_channel_id != -1); | |
49 } | |
50 | |
51 AudioSendStream::~AudioSendStream() { | |
52 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); | |
tommi
2015/10/16 11:15:33
does this go into the diagnostic log or are these
| |
53 } | |
54 | |
55 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | |
56 return webrtc::AudioSendStream::Stats(); | |
57 } | |
58 | |
59 void AudioSendStream::Start() { | |
60 } | |
61 | |
62 void AudioSendStream::Stop() { | |
63 } | |
64 | |
65 void AudioSendStream::SignalNetworkState(NetworkState state) { | |
66 } | |
67 | |
68 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { | |
69 return false; | |
70 } | |
71 } // namespace internal | |
72 } // namespace webrtc | |
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