solenberg | c7a8b08 | 2015-10-16 14:35:07 -0700 | [diff] [blame^] | 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(); |
| 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 |