Chromium Code Reviews| Index: webrtc/audio/audio_send_stream.cc | 
| diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..fcacce6dc51d8f4104d67a9a60355f015158d9bb | 
| --- /dev/null | 
| +++ b/webrtc/audio/audio_send_stream.cc | 
| @@ -0,0 +1,66 @@ | 
| +/* | 
| + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 
| + * | 
| + * Use of this source code is governed by a BSD-style license | 
| + * that can be found in the LICENSE file in the root of the source | 
| + * tree. An additional intellectual property rights grant can be found | 
| + * in the file PATENTS. All contributing project authors may | 
| + * be found in the AUTHORS file in the root of the source tree. | 
| + */ | 
| + | 
| +#include "webrtc/audio/audio_send_stream.h" | 
| + | 
| +#include <string> | 
| + | 
| +#include "webrtc/base/checks.h" | 
| + | 
| +namespace webrtc { | 
| +std::string AudioSendStream::Config::Rtp::ToString() const { | 
| 
 
tommi
2015/10/14 13:21:18
is this needed for debugging purposes? If so, can
 
the sun
2015/10/14 14:25:24
I like that idea, but it runs deeper than just thi
 
stefan-webrtc
2015/10/14 14:29:22
This string is used to LOG the stream configuratio
 
 | 
| + std::stringstream ss; | 
| + ss << "{ssrc: " << ssrc; | 
| + ss << ", extensions: ["; | 
| + for (size_t i = 0; i < extensions.size(); ++i) { | 
| 
 
tommi
2015/10/14 13:21:18
use range based loop?
 
the sun
2015/10/14 14:25:24
With the conditional "ss << ", ";" below, I don't
 
tommi
2015/10/14 14:49:28
Acknowledged.
 
 | 
| + ss << extensions[i].ToString(); | 
| + if (i != extensions.size() - 1) | 
| + ss << ", "; | 
| + } | 
| + ss << ']'; | 
| + ss << '}'; | 
| + return ss.str(); | 
| +} | 
| + | 
| +std::string AudioSendStream::Config::ToString() const { | 
| + std::stringstream ss; | 
| + ss << "{rtp: " << rtp.ToString(); | 
| + ss << ", voe_channel_id: " << voe_channel_id; | 
| + // TODO(solenberg): Encoder config. | 
| + ss << ", cng_payload_type: " << cng_payload_type; | 
| + ss << ", red_payload_type: " << red_payload_type; | 
| + ss << '}'; | 
| + return ss.str(); | 
| +} | 
| + | 
| +namespace internal { | 
| +AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config) | 
| + : config_(config) { | 
| + RTC_DCHECK(config.voe_channel_id != -1); | 
| +} | 
| + | 
| +webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { | 
| + return webrtc::AudioSendStream::Stats(); | 
| +} | 
| + | 
| +void AudioSendStream::Start() { | 
| +} | 
| + | 
| +void AudioSendStream::Stop() { | 
| +} | 
| + | 
| +void AudioSendStream::SignalNetworkState(NetworkState state) { | 
| +} | 
| + | 
| +bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { | 
| + return false; | 
| +} | 
| +} // namespace internal | 
| +} // namespace webrtc |