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_RECEIVE_STREAM_H_
12#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
13
14#include <map>
15#include <string>
16#include <vector>
17
18#include "webrtc/base/scoped_ptr.h"
19#include "webrtc/config.h"
20#include "webrtc/stream.h"
21#include "webrtc/transport.h"
22#include "webrtc/typedefs.h"
23
24namespace webrtc {
25
26class AudioDecoder;
27class AudioSinkInterface;
28
29// WORK IN PROGRESS
30// This class is under development and is not yet intended for for use outside
31// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
32// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
33
34class AudioReceiveStream : public ReceiveStream {
35 public:
36  struct Stats {
37    uint32_t remote_ssrc = 0;
38    int64_t bytes_rcvd = 0;
39    uint32_t packets_rcvd = 0;
40    uint32_t packets_lost = 0;
41    float fraction_lost = 0.0f;
42    std::string codec_name;
43    uint32_t ext_seqnum = 0;
44    uint32_t jitter_ms = 0;
45    uint32_t jitter_buffer_ms = 0;
46    uint32_t jitter_buffer_preferred_ms = 0;
47    uint32_t delay_estimate_ms = 0;
48    int32_t audio_level = -1;
49    float expand_rate = 0.0f;
50    float speech_expand_rate = 0.0f;
51    float secondary_decoded_rate = 0.0f;
52    float accelerate_rate = 0.0f;
53    float preemptive_expand_rate = 0.0f;
54    int32_t decoding_calls_to_silence_generator = 0;
55    int32_t decoding_calls_to_neteq = 0;
56    int32_t decoding_normal = 0;
57    int32_t decoding_plc = 0;
58    int32_t decoding_cng = 0;
59    int32_t decoding_plc_cng = 0;
60    int64_t capture_start_ntp_time_ms = 0;
61  };
62
63  struct Config {
64    std::string ToString() const;
65
66    // Receive-stream specific RTP settings.
67    struct Rtp {
68      std::string ToString() const;
69
70      // Synchronization source (stream identifier) to be received.
71      uint32_t remote_ssrc = 0;
72
73      // Sender SSRC used for sending RTCP (such as receiver reports).
74      uint32_t local_ssrc = 0;
75
76      // Enable feedback for send side bandwidth estimation.
77      // See
78      // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
79      // for details.
80      bool transport_cc = false;
81
82      // RTP header extensions used for the received stream.
83      std::vector<RtpExtension> extensions;
84    } rtp;
85
86    Transport* receive_transport = nullptr;
87    Transport* rtcp_send_transport = nullptr;
88
89    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
90    // level components.
91    // TODO(solenberg): Remove when VoiceEngine channels are created outside
92    // of Call.
93    int voe_channel_id = -1;
94
95    // Identifier for an A/V synchronization group. Empty string to disable.
96    // TODO(pbos): Synchronize streams in a sync group, not just one video
97    // stream to one audio stream. Tracked by issue webrtc:4762.
98    std::string sync_group;
99
100    // Decoders for every payload that we can receive. Call owns the
101    // AudioDecoder instances once the Config is submitted to
102    // Call::CreateReceiveStream().
103    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
104    std::map<uint8_t, AudioDecoder*> decoder_map;
105
106    // TODO(pbos): Remove config option once combined A/V BWE is always on.
107    bool combined_audio_video_bwe = false;
108  };
109
110  virtual Stats GetStats() const = 0;
111
112  // Sets an audio sink that receives unmixed audio from the receive stream.
113  // Ownership of the sink is passed to the stream and can be used by the
114  // caller to do lifetime management (i.e. when the sink's dtor is called).
115  // Only one sink can be set and passing a null sink, clears an existing one.
116  // NOTE: Audio must still somehow be pulled through AudioTransport for audio
117  // to stream through this sink. In practice, this happens if mixed audio
118  // is being pulled+rendered and/or if audio is being pulled for the purposes
119  // of feeding to the AEC.
120  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
121};
122}  // namespace webrtc
123
124#endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_
125