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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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