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

Issue 1390753002: Implement AudioReceiveStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: merge master Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include "webrtc/audio_receive_stream.h" 14 #include "webrtc/audio_receive_stream.h"
15 #include "webrtc/audio/scoped_voe_interface.h"
16 #include "webrtc/base/thread_checker.h"
15 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
18 #include "webrtc/voice_engine/include/voe_base.h"
16 19
17 namespace webrtc { 20 namespace webrtc {
18 21
19 class RemoteBitrateEstimator; 22 class RemoteBitrateEstimator;
23 class VoiceEngine;
20 24
21 namespace internal { 25 namespace internal {
22 26
23 class AudioReceiveStream : public webrtc::AudioReceiveStream { 27 class AudioReceiveStream : public webrtc::AudioReceiveStream {
24 public: 28 public:
25 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, 29 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
26 const webrtc::AudioReceiveStream::Config& config); 30 const webrtc::AudioReceiveStream::Config& config,
31 VoiceEngine* voice_engine);
27 ~AudioReceiveStream() override; 32 ~AudioReceiveStream() override;
28 33
29 // webrtc::ReceiveStream implementation. 34 // webrtc::ReceiveStream implementation.
30 void Start() override; 35 void Start() override;
31 void Stop() override; 36 void Stop() override;
32 void SignalNetworkState(NetworkState state) override; 37 void SignalNetworkState(NetworkState state) override;
33 bool DeliverRtcp(const uint8_t* packet, size_t length) override; 38 bool DeliverRtcp(const uint8_t* packet, size_t length) override;
34 bool DeliverRtp(const uint8_t* packet, 39 bool DeliverRtp(const uint8_t* packet,
35 size_t length, 40 size_t length,
36 const PacketTime& packet_time) override; 41 const PacketTime& packet_time) override;
37 42
38 // webrtc::AudioReceiveStream implementation. 43 // webrtc::AudioReceiveStream implementation.
39 webrtc::AudioReceiveStream::Stats GetStats() const override; 44 webrtc::AudioReceiveStream::Stats GetStats() const override;
40 45
41 const webrtc::AudioReceiveStream::Config& config() const; 46 const webrtc::AudioReceiveStream::Config& config() const;
42 47
43 private: 48 private:
49 rtc::ThreadChecker thread_checker_;
44 RemoteBitrateEstimator* const remote_bitrate_estimator_; 50 RemoteBitrateEstimator* const remote_bitrate_estimator_;
45 const webrtc::AudioReceiveStream::Config config_; 51 const webrtc::AudioReceiveStream::Config config_;
52 VoiceEngine* voice_engine_;
53 // We hold one interface pointer to the VoE to make sure it is kept alive.
54 ScopedVoEInterface<VoEBase> voe_base_;
46 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; 55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
47 }; 56 };
48 } // namespace internal 57 } // namespace internal
49 } // namespace webrtc 58 } // namespace webrtc
50 59
51 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 60 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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