Line data Source code
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_CALL_AUDIO_RECEIVE_STREAM_H_
12 : #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
13 :
14 : #include <map>
15 : #include <memory>
16 : #include <string>
17 : #include <vector>
18 :
19 : #include "webrtc/api/call/transport.h"
20 : #include "webrtc/base/optional.h"
21 : #include "webrtc/base/scoped_ref_ptr.h"
22 : #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
23 : #include "webrtc/common_types.h"
24 : #include "webrtc/config.h"
25 : #include "webrtc/typedefs.h"
26 :
27 : namespace webrtc {
28 : class AudioSinkInterface;
29 :
30 : // WORK IN PROGRESS
31 : // This class is under development and is not yet intended for for use outside
32 : // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
33 : // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
34 :
35 0 : class AudioReceiveStream {
36 : public:
37 0 : struct Stats {
38 : uint32_t remote_ssrc = 0;
39 : int64_t bytes_rcvd = 0;
40 : uint32_t packets_rcvd = 0;
41 : uint32_t packets_lost = 0;
42 : float fraction_lost = 0.0f;
43 : std::string codec_name;
44 : rtc::Optional<int> codec_payload_type;
45 : uint32_t ext_seqnum = 0;
46 : uint32_t jitter_ms = 0;
47 : uint32_t jitter_buffer_ms = 0;
48 : uint32_t jitter_buffer_preferred_ms = 0;
49 : uint32_t delay_estimate_ms = 0;
50 : int32_t audio_level = -1;
51 : float expand_rate = 0.0f;
52 : float speech_expand_rate = 0.0f;
53 : float secondary_decoded_rate = 0.0f;
54 : float accelerate_rate = 0.0f;
55 : float preemptive_expand_rate = 0.0f;
56 : int32_t decoding_calls_to_silence_generator = 0;
57 : int32_t decoding_calls_to_neteq = 0;
58 : int32_t decoding_normal = 0;
59 : int32_t decoding_plc = 0;
60 : int32_t decoding_cng = 0;
61 : int32_t decoding_plc_cng = 0;
62 : int32_t decoding_muted_output = 0;
63 : int64_t capture_start_ntp_time_ms = 0;
64 : };
65 :
66 0 : struct Config {
67 : std::string ToString() const;
68 :
69 : // Receive-stream specific RTP settings.
70 0 : struct Rtp {
71 : std::string ToString() const;
72 :
73 : // Synchronization source (stream identifier) to be received.
74 : uint32_t remote_ssrc = 0;
75 :
76 : // Sender SSRC used for sending RTCP (such as receiver reports).
77 : uint32_t local_ssrc = 0;
78 :
79 : // Enable feedback for send side bandwidth estimation.
80 : // See
81 : // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
82 : // for details.
83 : bool transport_cc = false;
84 :
85 : // See NackConfig for description.
86 : NackConfig nack;
87 :
88 : // RTP header extensions used for the received stream.
89 : std::vector<RtpExtension> extensions;
90 : } rtp;
91 :
92 : Transport* rtcp_send_transport = nullptr;
93 :
94 : // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
95 : // level components.
96 : // TODO(solenberg): Remove when VoiceEngine channels are created outside
97 : // of Call.
98 : int voe_channel_id = -1;
99 :
100 : // Identifier for an A/V synchronization group. Empty string to disable.
101 : // TODO(pbos): Synchronize streams in a sync group, not just one video
102 : // stream to one audio stream. Tracked by issue webrtc:4762.
103 : std::string sync_group;
104 :
105 : // Decoders for every payload that we can receive. Call owns the
106 : // AudioDecoder instances once the Config is submitted to
107 : // Call::CreateReceiveStream().
108 : // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
109 : std::map<uint8_t, AudioDecoder*> decoder_map;
110 :
111 : rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
112 : };
113 :
114 : // Starts stream activity.
115 : // When a stream is active, it can receive, process and deliver packets.
116 : virtual void Start() = 0;
117 : // Stops stream activity.
118 : // When a stream is stopped, it can't receive, process or deliver packets.
119 : virtual void Stop() = 0;
120 :
121 : virtual Stats GetStats() const = 0;
122 :
123 : // Sets an audio sink that receives unmixed audio from the receive stream.
124 : // Ownership of the sink is passed to the stream and can be used by the
125 : // caller to do lifetime management (i.e. when the sink's dtor is called).
126 : // Only one sink can be set and passing a null sink clears an existing one.
127 : // NOTE: Audio must still somehow be pulled through AudioTransport for audio
128 : // to stream through this sink. In practice, this happens if mixed audio
129 : // is being pulled+rendered and/or if audio is being pulled for the purposes
130 : // of feeding to the AEC.
131 : virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
132 :
133 : // Sets playback gain of the stream, applied when mixing, and thus after it
134 : // is potentially forwarded to any attached AudioSinkInterface implementation.
135 : virtual void SetGain(float gain) = 0;
136 :
137 : protected:
138 0 : virtual ~AudioReceiveStream() {}
139 : };
140 : } // namespace webrtc
141 :
142 : #endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
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