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_AUDIO_SEND_STREAM_H_
12#define WEBRTC_AUDIO_SEND_STREAM_H_
13
14#include <string>
15#include <vector>
16
17#include "webrtc/base/scoped_ptr.h"
18#include "webrtc/config.h"
19#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
20#include "webrtc/stream.h"
21#include "webrtc/transport.h"
22#include "webrtc/typedefs.h"
23
24namespace 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
31class AudioSendStream : public SendStream {
32 public:
33  struct Stats {
34    // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
35    uint32_t local_ssrc = 0;
36    int64_t bytes_sent = 0;
37    int32_t packets_sent = 0;
38    int32_t packets_lost = -1;
39    float fraction_lost = -1.0f;
40    std::string codec_name;
41    int32_t ext_seqnum = -1;
42    int32_t jitter_ms = -1;
43    int64_t rtt_ms = -1;
44    int32_t audio_level = -1;
45    float aec_quality_min = -1.0f;
46    int32_t echo_delay_median_ms = -1;
47    int32_t echo_delay_std_ms = -1;
48    int32_t echo_return_loss = -100;
49    int32_t echo_return_loss_enhancement = -100;
50    bool typing_noise_detected = false;
51  };
52
53  struct Config {
54    Config() = delete;
55    explicit Config(Transport* send_transport)
56        : send_transport(send_transport) {}
57
58    std::string ToString() const;
59
60    // Receive-stream specific RTP settings.
61    struct Rtp {
62      std::string ToString() const;
63
64      // Sender SSRC.
65      uint32_t ssrc = 0;
66
67      // RTP header extensions used for the sent stream.
68      std::vector<RtpExtension> extensions;
69
70      // RTCP CNAME, see RFC 3550.
71      std::string c_name;
72    } rtp;
73
74    // Transport for outgoing packets. The transport is expected to exist for
75    // the entire life of the AudioSendStream and is owned by the API client.
76    Transport* send_transport = nullptr;
77
78    // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
79    // components.
80    // TODO(solenberg): Remove when VoiceEngine channels are created outside
81    // of Call.
82    int voe_channel_id = -1;
83
84    // Ownership of the encoder object is transferred to Call when the config is
85    // passed to Call::CreateAudioSendStream().
86    // TODO(solenberg): Implement, once we configure codecs through the new API.
87    // rtc::scoped_ptr<AudioEncoder> encoder;
88    int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
89    int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
90  };
91
92  // TODO(solenberg): Make payload_type a config property instead.
93  virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
94                                  uint32_t duration_ms) = 0;
95  virtual Stats GetStats() const = 0;
96};
97}  // namespace webrtc
98
99#endif  // WEBRTC_AUDIO_SEND_STREAM_H_
100