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Side by Side Diff: webrtc/audio_send_stream.h

Issue 1226123005: Define Stream base classes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Code review follow-up 3 Created 5 years, 5 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_SEND_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_SEND_STREAM_H_
12 #define WEBRTC_AUDIO_SEND_STREAM_H_ 12 #define WEBRTC_AUDIO_SEND_STREAM_H_
13 13
14 #include <string> 14 #include <string>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/base/scoped_ptr.h" 17 #include "webrtc/base/scoped_ptr.h"
18 #include "webrtc/config.h" 18 #include "webrtc/config.h"
19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" 19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
20 #include "webrtc/stream.h"
20 #include "webrtc/typedefs.h" 21 #include "webrtc/typedefs.h"
21 22
22 namespace webrtc { 23 namespace webrtc {
23 24
24 class AudioSendStream { 25 class AudioSendStream : public SendStream {
25 public: 26 public:
26 struct Stats {}; 27 struct Stats {};
27 28
28 struct Config { 29 struct Config {
29 std::string ToString() const; 30 std::string ToString() const;
30 31
31 // Receive-stream specific RTP settings. 32 // Receive-stream specific RTP settings.
32 struct Rtp { 33 struct Rtp {
33 std::string ToString() const; 34 std::string ToString() const;
34 35
35 // Sender SSRC. 36 // Sender SSRC.
36 uint32_t ssrc = 0; 37 uint32_t ssrc = 0;
37 38
38 // RTP header extensions used for the received stream. 39 // RTP header extensions used for the received stream.
39 std::vector<RtpExtension> extensions; 40 std::vector<RtpExtension> extensions;
40 } rtp; 41 } rtp;
41 42
42 rtc::scoped_ptr<AudioEncoder> encoder; 43 rtc::scoped_ptr<AudioEncoder> encoder;
43 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. 44 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
44 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. 45 int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
45 }; 46 };
46 47
47 virtual Stats GetStats() const = 0; 48 virtual Stats GetStats() const = 0;
48
49 protected:
50 virtual ~AudioSendStream() {}
51 }; 49 };
52 } // namespace webrtc 50 } // namespace webrtc
53 51
54 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ 52 #endif // WEBRTC_AUDIO_SEND_STREAM_H_
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