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 |