1 /*
2  *  Copyright (c) 2012 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_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
13 
14 #include "webrtc/common_types.h"
15 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
16 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
18 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
19 #include "webrtc/typedefs.h"
20 
21 namespace webrtc {
22 class RTPSenderAudio : public DTMFqueue {
23  public:
24   RTPSenderAudio(Clock* clock,
25                  RTPSender* rtpSender,
26                  RtpAudioFeedback* audio_feedback);
27   virtual ~RTPSenderAudio();
28 
29   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
30                                int8_t payloadType,
31                                uint32_t frequency,
32                                size_t channels,
33                                uint32_t rate,
34                                RtpUtility::Payload** payload);
35 
36   int32_t SendAudio(FrameType frameType,
37                     int8_t payloadType,
38                     uint32_t captureTimeStamp,
39                     const uint8_t* payloadData,
40                     size_t payloadSize,
41                     const RTPFragmentationHeader* fragmentation);
42 
43   // set audio packet size, used to determine when it's time to send a DTMF
44   // packet in silence (CNG)
45   int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
46 
47   // Store the audio level in dBov for
48   // header-extension-for-audio-level-indication.
49   // Valid range is [0,100]. Actual value is negative.
50   int32_t SetAudioLevel(uint8_t level_dBov);
51 
52   // Send a DTMF tone using RFC 2833 (4733)
53   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
54 
55   int AudioFrequency() const;
56 
57   // Set payload type for Redundant Audio Data RFC 2198
58   int32_t SetRED(int8_t payloadType);
59 
60   // Get payload type for Redundant Audio Data RFC 2198
61   int32_t RED(int8_t* payloadType) const;
62 
63  protected:
64   int32_t SendTelephoneEventPacket(
65       bool ended,
66       int8_t dtmf_payload_type,
67       uint32_t dtmfTimeStamp,
68       uint16_t duration,
69       bool markerBit);  // set on first packet in talk burst
70 
71   bool MarkerBit(const FrameType frameType, const int8_t payloadType);
72 
73  private:
74   Clock* const _clock;
75   RTPSender* const _rtpSender;
76   RtpAudioFeedback* const _audioFeedback;
77 
78   rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
79 
80   uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
81 
82   // DTMF
83   bool _dtmfEventIsOn;
84   bool _dtmfEventFirstPacketSent;
85   int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
86   uint32_t _dtmfTimestamp;
87   uint8_t _dtmfKey;
88   uint32_t _dtmfLengthSamples;
89   uint8_t _dtmfLevel;
90   int64_t _dtmfTimeLastSent;
91   uint32_t _dtmfTimestampLastSent;
92 
93   int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
94 
95   // VAD detection, used for markerbit
96   bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
97   int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
98   int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
99   int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
100   int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
101   int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
102 
103   // Audio level indication
104   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
105   uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
106 };
107 }  // namespace webrtc
108 
109 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
110