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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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