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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/voip/voip_dtmf.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/voip/voip_dtmf.h')
-rw-r--r-- | third_party/libwebrtc/api/voip/voip_dtmf.h | 74 |
1 files changed, 74 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/voip_dtmf.h b/third_party/libwebrtc/api/voip/voip_dtmf.h new file mode 100644 index 0000000000..ef7ea28c94 --- /dev/null +++ b/third_party/libwebrtc/api/voip/voip_dtmf.h @@ -0,0 +1,74 @@ +/* + * 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. + // Returns following VoipResult; + // kOk - telephone event type is registered as provided. + // kInvalidArgument - `channel_id` is invalid. + virtual VoipResult 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 following VoipResult; + // kOk - requested DTMF event is successfully scheduled. + // kInvalidArgument - `channel_id` is invalid. + // kFailedPrecondition - Missing prerequisite on RegisterTelephoneEventType + // or sending state. + virtual VoipResult SendDtmfEvent(ChannelId channel_id, + DtmfEvent dtmf_event, + int duration_ms) = 0; + + protected: + virtual ~VoipDtmf() = default; +}; + +} // namespace webrtc + +#endif // API_VOIP_VOIP_DTMF_H_ |