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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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