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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/stream.h" |
21 #include "webrtc/transport.h" | 21 #include "webrtc/transport.h" |
22 #include "webrtc/typedefs.h" | 22 #include "webrtc/typedefs.h" |
23 | 23 |
24 namespace webrtc { | 24 namespace webrtc { |
25 | 25 |
26 class AudioSendStream : public SendStream { | 26 class AudioSendStream : public SendStream { |
27 public: | 27 public: |
28 struct Stats {}; | 28 struct Stats { |
| 29 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. |
| 30 uint32_t local_ssrc = 0; |
| 31 int64_t bytes_sent = 0; |
| 32 int32_t packets_sent = 0; |
| 33 int32_t packets_lost = -1; |
| 34 float fraction_lost = -1.0f; |
| 35 std::string codec_name; |
| 36 int32_t ext_seqnum = -1; |
| 37 int32_t jitter_ms = -1; |
| 38 int64_t rtt_ms = -1; |
| 39 int32_t audio_level = -1; |
| 40 float aec_quality_min = -1.0f; |
| 41 int32_t echo_delay_median_ms = -1; |
| 42 int32_t echo_delay_std_ms = -1; |
| 43 int32_t echo_return_loss = -100; |
| 44 int32_t echo_return_loss_enhancement = -100; |
| 45 bool typing_noise_detected = false; |
| 46 }; |
29 | 47 |
30 struct Config { | 48 struct Config { |
31 Config() = delete; | 49 Config() = delete; |
32 explicit Config(Transport* send_transport) | 50 explicit Config(Transport* send_transport) |
33 : send_transport(send_transport) {} | 51 : send_transport(send_transport) {} |
34 | 52 |
35 std::string ToString() const; | 53 std::string ToString() const; |
36 | 54 |
37 // Receive-stream specific RTP settings. | 55 // Receive-stream specific RTP settings. |
38 struct Rtp { | 56 struct Rtp { |
(...skipping 22 matching lines...) Expand all Loading... |
61 // rtc::scoped_ptr<AudioEncoder> encoder; | 79 // rtc::scoped_ptr<AudioEncoder> encoder; |
62 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. | 80 int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. |
63 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. | 81 int red_payload_type = -1; // pt, or -1 to disable REDundant coding. |
64 }; | 82 }; |
65 | 83 |
66 virtual Stats GetStats() const = 0; | 84 virtual Stats GetStats() const = 0; |
67 }; | 85 }; |
68 } // namespace webrtc | 86 } // namespace webrtc |
69 | 87 |
70 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ | 88 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ |
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