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

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years 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_state.h" 15 #include "webrtc/audio_state.h"
16 #include "webrtc/base/thread_checker.h" 16 #include "webrtc/base/thread_checker.h"
17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
18 18
19 namespace webrtc { 19 namespace webrtc {
20 class RemoteBitrateEstimator; 20 class RemoteBitrateEstimator;
21 21
22 namespace voe { 22 namespace voe {
23 class ChannelProxy; 23 class ChannelProxy;
24 } // namespace voe 24 } // namespace voe
25 25
26 namespace internal { 26 namespace internal {
27
27 class AudioReceiveStream final : public webrtc::AudioReceiveStream { 28 class AudioReceiveStream final : public webrtc::AudioReceiveStream {
28 public: 29 public:
29 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, 30 AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
30 const webrtc::AudioReceiveStream::Config& config, 31 const webrtc::AudioReceiveStream::Config& config,
31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state); 32 const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
32 ~AudioReceiveStream() override; 33 ~AudioReceiveStream() override;
33 34
34 // webrtc::ReceiveStream implementation. 35 // webrtc::ReceiveStream implementation.
35 void Start() override; 36 void Start() override;
36 void Stop() override; 37 void Stop() override;
37 void SignalNetworkState(NetworkState state) override; 38 void SignalNetworkState(NetworkState state) override;
38 bool DeliverRtcp(const uint8_t* packet, size_t length) override; 39 bool DeliverRtcp(const uint8_t* packet, size_t length) override;
39 bool DeliverRtp(const uint8_t* packet, 40 bool DeliverRtp(const uint8_t* packet,
40 size_t length, 41 size_t length,
41 const PacketTime& packet_time) override; 42 const PacketTime& packet_time) override;
42 43
43 // webrtc::AudioReceiveStream implementation. 44 // webrtc::AudioReceiveStream implementation.
44 webrtc::AudioReceiveStream::Stats GetStats() const override; 45 webrtc::AudioReceiveStream::Stats GetStats() const override;
45 46
47 void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
48
46 const webrtc::AudioReceiveStream::Config& config() const; 49 const webrtc::AudioReceiveStream::Config& config() const;
47 50
48 private: 51 private:
49 VoiceEngine* voice_engine() const; 52 VoiceEngine* voice_engine() const;
50 53
51 rtc::ThreadChecker thread_checker_; 54 rtc::ThreadChecker thread_checker_;
52 RemoteBitrateEstimator* const remote_bitrate_estimator_; 55 RemoteBitrateEstimator* const remote_bitrate_estimator_;
53 const webrtc::AudioReceiveStream::Config config_; 56 const webrtc::AudioReceiveStream::Config config_;
54 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 57 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
55 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; 58 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
56 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_; 59 rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
57 60
58 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); 61 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
59 }; 62 };
60 } // namespace internal 63 } // namespace internal
61 } // namespace webrtc 64 } // namespace webrtc
62 65
63 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 66 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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