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Side by Side Diff: webrtc/call/audio_send_stream.h

Issue 3004723002: Move RtpExtension to api/ directory and config.h/.cc to call/. (Closed)
Patch Set: Fix nit. Created 3 years, 3 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_CALL_AUDIO_SEND_STREAM_H_ 11 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
12 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_ 12 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/api/audio_codecs/audio_encoder_factory.h" 18 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
19 #include "webrtc/api/audio_codecs/audio_format.h" 19 #include "webrtc/api/audio_codecs/audio_format.h"
20 #include "webrtc/api/call/transport.h" 20 #include "webrtc/api/call/transport.h"
21 #include "webrtc/config.h" 21 #include "webrtc/api/rtpparameters.h"
22 #include "webrtc/call/rtp_config.h"
22 #include "webrtc/rtc_base/optional.h" 23 #include "webrtc/rtc_base/optional.h"
24 #include "webrtc/rtc_base/scoped_ref_ptr.h"
23 #include "webrtc/typedefs.h" 25 #include "webrtc/typedefs.h"
24 26
25 namespace webrtc { 27 namespace webrtc {
26 28
27 // WORK IN PROGRESS 29 // WORK IN PROGRESS
28 // This class is under development and is not yet intended for for use outside 30 // This class is under development and is not yet intended for for use outside
29 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. 31 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
30 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 32 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
31 33
32 class AudioSendStream { 34 class AudioSendStream {
(...skipping 114 matching lines...) Expand 10 before | Expand all | Expand 10 after
147 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, 149 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
148 int event, int duration_ms) = 0; 150 int event, int duration_ms) = 0;
149 151
150 virtual void SetMuted(bool muted) = 0; 152 virtual void SetMuted(bool muted) = 0;
151 153
152 virtual Stats GetStats() const = 0; 154 virtual Stats GetStats() const = 0;
153 }; 155 };
154 } // namespace webrtc 156 } // namespace webrtc
155 157
156 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_ 158 #endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_
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