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Side by Side Diff: webrtc/audio/audio_send_stream.h

Issue 1397123003: Add AudioSendStream to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 months ago
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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 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
13
14 #include "webrtc/audio_send_stream.h"
15
16 namespace webrtc {
17 namespace internal {
18
19 class AudioSendStream : public webrtc::AudioSendStream {
20 public:
21 explicit AudioSendStream(const webrtc::AudioSendStream::Config& config);
22 ~AudioSendStream() override;
23
24 // webrtc::SendStream implementation.
25 void Start() override;
26 void Stop() override;
27 void SignalNetworkState(NetworkState state) override;
28 bool DeliverRtcp(const uint8_t* packet, size_t length) override;
29
30 // webrtc::AudioSendStream implementation.
31 webrtc::AudioSendStream::Stats GetStats() const override;
32
33 const webrtc::AudioSendStream::Config& config() const {
34 return config_;
35 }
36
37 private:
38 const webrtc::AudioSendStream::Config config_;
39 };
40 } // namespace internal
41 } // namespace webrtc
42
43 #endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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