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Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 1728503002: Replace scoped_ptr with unique_ptr in webrtc/media/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up1
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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_MEDIA_BASE_MEDIACHANNEL_H_ 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
13 13
14 #include <memory>
14 #include <string> 15 #include <string>
15 #include <vector> 16 #include <vector>
16 17
17 #include "webrtc/base/basictypes.h" 18 #include "webrtc/base/basictypes.h"
18 #include "webrtc/base/buffer.h" 19 #include "webrtc/base/buffer.h"
19 #include "webrtc/base/dscp.h" 20 #include "webrtc/base/dscp.h"
20 #include "webrtc/base/logging.h" 21 #include "webrtc/base/logging.h"
21 #include "webrtc/base/optional.h" 22 #include "webrtc/base/optional.h"
22 #include "webrtc/base/sigslot.h" 23 #include "webrtc/base/sigslot.h"
23 #include "webrtc/base/socket.h" 24 #include "webrtc/base/socket.h"
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921 // Send a DTMF |event|. The DTMF out-of-band signal will be used. 922 // Send a DTMF |event|. The DTMF out-of-band signal will be used.
922 // The |ssrc| should be either 0 or a valid send stream ssrc. 923 // The |ssrc| should be either 0 or a valid send stream ssrc.
923 // The valid value for the |event| are 0 to 15 which corresponding to 924 // The valid value for the |event| are 0 to 15 which corresponding to
924 // DTMF event 0-9, *, #, A-D. 925 // DTMF event 0-9, *, #, A-D.
925 virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0; 926 virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
926 // Gets quality stats for the channel. 927 // Gets quality stats for the channel.
927 virtual bool GetStats(VoiceMediaInfo* info) = 0; 928 virtual bool GetStats(VoiceMediaInfo* info) = 0;
928 929
929 virtual void SetRawAudioSink( 930 virtual void SetRawAudioSink(
930 uint32_t ssrc, 931 uint32_t ssrc,
931 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0; 932 std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
932 }; 933 };
933 934
934 struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> { 935 struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
935 // Use conference mode? This flag comes from the remote 936 // Use conference mode? This flag comes from the remote
936 // description's SDP line 'a=x-google-flag:conference', copied over 937 // description's SDP line 'a=x-google-flag:conference', copied over
937 // by VideoChannel::SetRemoteContent_w, and ultimately used by 938 // by VideoChannel::SetRemoteContent_w, and ultimately used by
938 // conference mode screencast logic in 939 // conference mode screencast logic in
939 // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig. 940 // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
940 // The special screencast behaviour is disabled by default. 941 // The special screencast behaviour is disabled by default.
941 bool conference_mode = false; 942 bool conference_mode = false;
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1107 // Signal when the media channel is ready to send the stream. Arguments are: 1108 // Signal when the media channel is ready to send the stream. Arguments are:
1108 // writable(bool) 1109 // writable(bool)
1109 sigslot::signal1<bool> SignalReadyToSend; 1110 sigslot::signal1<bool> SignalReadyToSend;
1110 // Signal for notifying that the remote side has closed the DataChannel. 1111 // Signal for notifying that the remote side has closed the DataChannel.
1111 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1112 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1112 }; 1113 };
1113 1114
1114 } // namespace cricket 1115 } // namespace cricket
1115 1116
1116 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1117 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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