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Side by Side Diff: webrtc/video/audio_receive_stream.h

Issue 1227923005: Split webrtc/video into webrtc/{audio,call,video}. (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_VIDEO_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
13
14 #include "webrtc/audio_receive_stream.h"
15 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
16
17 namespace webrtc {
18
19 class RemoteBitrateEstimator;
20
21 namespace internal {
22
23 class AudioReceiveStream : public webrtc::AudioReceiveStream {
24 public:
25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
26 const webrtc::AudioReceiveStream::Config& config);
27 ~AudioReceiveStream() override {}
28
29 // webrtc::ReceiveStream implementation.
30 void Start() override;
31 void Stop() override;
32 void SignalNetworkState(NetworkState state) override;
33 bool DeliverRtcp(const uint8_t* packet, size_t length) override;
34 bool DeliverRtp(const uint8_t* packet,
35 size_t length,
36 const PacketTime& packet_time) override;
37
38 // webrtc::AudioReceiveStream implementation.
39 webrtc::AudioReceiveStream::Stats GetStats() const override;
40
41 const webrtc::AudioReceiveStream::Config& config() const {
42 return config_;
43 }
44
45 private:
46 RemoteBitrateEstimator* const remote_bitrate_estimator_;
47 const webrtc::AudioReceiveStream::Config config_;
48 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
49 };
50 } // namespace internal
51 } // namespace webrtc
52
53 #endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
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