| /* |
| * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef API_VOIP_VOIP_DTMF_H_ |
| #define API_VOIP_VOIP_DTMF_H_ |
| |
| #include "api/voip/voip_base.h" |
| |
| namespace webrtc { |
| |
| // DTMF events and their event codes as defined in |
| // https://tools.ietf.org/html/rfc4733#section-7 |
| enum class DtmfEvent : uint8_t { |
| kDigitZero = 0, |
| kDigitOne, |
| kDigitTwo, |
| kDigitThree, |
| kDigitFour, |
| kDigitFive, |
| kDigitSix, |
| kDigitSeven, |
| kDigitEight, |
| kDigitNine, |
| kAsterisk, |
| kHash, |
| kLetterA, |
| kLetterB, |
| kLetterC, |
| kLetterD |
| }; |
| |
| // VoipDtmf interface provides DTMF related interfaces such |
| // as sending DTMF events to the remote endpoint. |
| class VoipDtmf { |
| public: |
| // Register the payload type and sample rate for DTMF (RFC 4733) payload. |
| // Must be called exactly once prior to calling SendDtmfEvent after payload |
| // type has been negotiated with remote. |
| virtual void RegisterTelephoneEventType(ChannelId channel_id, |
| int rtp_payload_type, |
| int sample_rate_hz) = 0; |
| |
| // Send DTMF named event as specified by |
| // https://tools.ietf.org/html/rfc4733#section-3.2 |
| // |duration_ms| specifies the duration of DTMF packets that will be emitted |
| // in place of real RTP packets instead. |
| // Must be called after RegisterTelephoneEventType and VoipBase::StartSend |
| // have been called. |
| // Returns true if the requested DTMF event is successfully scheduled. |
| virtual bool SendDtmfEvent(ChannelId channel_id, |
| DtmfEvent dtmf_event, |
| int duration_ms) = 0; |
| |
| protected: |
| virtual ~VoipDtmf() = default; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_VOIP_VOIP_DTMF_H_ |