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1 /* | |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | |
12 #define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | |
13 | |
14 #include <memory> | |
15 #include <string> | |
16 #include <vector> | |
17 | |
18 #include "webrtc/api/call/transport.h" | |
19 #include "webrtc/base/optional.h" | |
20 #include "webrtc/config.h" | |
21 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | |
22 #include "webrtc/typedefs.h" | |
23 | |
24 namespace webrtc { | |
25 | |
26 // WORK IN PROGRESS | |
27 // This class is under development and is not yet intended for for use outside | |
28 // of WebRtc/Libjingle. Please use the VoiceEngine API instead. | |
29 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690 | |
30 | |
31 class AudioSendStream { | |
32 public: | |
33 struct Stats { | |
34 Stats(); | |
35 ~Stats(); | |
36 | |
37 // TODO(solenberg): Harmonize naming and defaults with receive stream stats. | |
38 uint32_t local_ssrc = 0; | |
39 int64_t bytes_sent = 0; | |
40 int32_t packets_sent = 0; | |
41 int32_t packets_lost = -1; | |
42 float fraction_lost = -1.0f; | |
43 std::string codec_name; | |
44 rtc::Optional<int> codec_payload_type; | |
45 int32_t ext_seqnum = -1; | |
46 int32_t jitter_ms = -1; | |
47 int64_t rtt_ms = -1; | |
48 int32_t audio_level = -1; | |
49 float aec_quality_min = -1.0f; | |
50 int32_t echo_delay_median_ms = -1; | |
51 int32_t echo_delay_std_ms = -1; | |
52 int32_t echo_return_loss = -100; | |
53 int32_t echo_return_loss_enhancement = -100; | |
54 float residual_echo_likelihood = -1.0f; | |
55 bool typing_noise_detected = false; | |
56 }; | |
57 | |
58 struct Config { | |
59 Config() = delete; | |
60 explicit Config(Transport* send_transport); | |
61 ~Config(); | |
62 std::string ToString() const; | |
63 | |
64 // Send-stream specific RTP settings. | |
65 struct Rtp { | |
66 Rtp(); | |
67 ~Rtp(); | |
68 std::string ToString() const; | |
69 | |
70 // Sender SSRC. | |
71 uint32_t ssrc = 0; | |
72 | |
73 // RTP header extensions used for the sent stream. | |
74 std::vector<RtpExtension> extensions; | |
75 | |
76 // See NackConfig for description. | |
77 NackConfig nack; | |
78 | |
79 // RTCP CNAME, see RFC 3550. | |
80 std::string c_name; | |
81 } rtp; | |
82 | |
83 // Transport for outgoing packets. The transport is expected to exist for | |
84 // the entire life of the AudioSendStream and is owned by the API client. | |
85 Transport* send_transport = nullptr; | |
86 | |
87 // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level | |
88 // components. | |
89 // TODO(solenberg): Remove when VoiceEngine channels are created outside | |
90 // of Call. | |
91 int voe_channel_id = -1; | |
92 | |
93 // Bitrate limits used for variable audio bitrate streams. Set both to -1 to | |
94 // disable audio bitrate adaptation. | |
95 // Note: This is still an experimental feature and not ready for real usage. | |
96 int min_bitrate_bps = -1; | |
97 int max_bitrate_bps = -1; | |
98 | |
99 // Defines whether to turn on audio network adaptor, and defines its config | |
100 // string. | |
101 rtc::Optional<std::string> audio_network_adaptor_config; | |
102 | |
103 struct SendCodecSpec { | |
104 SendCodecSpec(); | |
105 std::string ToString() const; | |
106 | |
107 bool operator==(const SendCodecSpec& rhs) const; | |
108 bool operator!=(const SendCodecSpec& rhs) const { | |
109 return !(*this == rhs); | |
110 } | |
111 | |
112 bool nack_enabled = false; | |
113 bool transport_cc_enabled = false; | |
114 bool enable_codec_fec = false; | |
115 bool enable_opus_dtx = false; | |
116 int opus_max_playback_rate = 0; | |
117 int cng_payload_type = -1; | |
118 int cng_plfreq = -1; | |
119 int max_ptime_ms = -1; | |
120 int min_ptime_ms = -1; | |
121 webrtc::CodecInst codec_inst; | |
122 } send_codec_spec; | |
123 }; | |
124 | |
125 // Starts stream activity. | |
126 // When a stream is active, it can receive, process and deliver packets. | |
127 virtual void Start() = 0; | |
128 // Stops stream activity. | |
129 // When a stream is stopped, it can't receive, process or deliver packets. | |
130 virtual void Stop() = 0; | |
131 | |
132 // TODO(solenberg): Make payload_type a config property instead. | |
133 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency, | |
134 int event, int duration_ms) = 0; | |
135 | |
136 virtual void SetMuted(bool muted) = 0; | |
137 | |
138 virtual Stats GetStats() const = 0; | |
139 | |
140 protected: | |
141 virtual ~AudioSendStream() {} | |
142 }; | |
143 } // namespace webrtc | |
144 | |
145 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ | |
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