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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/api/dtmf_sender_interface.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/dtmf_sender_interface.h')
-rw-r--r-- | third_party/libwebrtc/api/dtmf_sender_interface.h | 125 |
1 files changed, 125 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/dtmf_sender_interface.h b/third_party/libwebrtc/api/dtmf_sender_interface.h new file mode 100644 index 0000000000..d63e66bbf7 --- /dev/null +++ b/third_party/libwebrtc/api/dtmf_sender_interface.h @@ -0,0 +1,125 @@ +/* + * Copyright 2012 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_DTMF_SENDER_INTERFACE_H_ +#define API_DTMF_SENDER_INTERFACE_H_ + +#include <string> + +#include "api/media_stream_interface.h" +#include "rtc_base/ref_count.h" + +namespace webrtc { + +// DtmfSender callback interface, used to implement RTCDtmfSender events. +// Applications should implement this interface to get notifications from the +// DtmfSender. +class DtmfSenderObserverInterface { + public: + // Triggered when DTMF `tone` is sent. + // If `tone` is empty that means the DtmfSender has sent out all the given + // tones. + // The callback includes the state of the tone buffer at the time when + // the tone finished playing. + virtual void OnToneChange(const std::string& tone, + const std::string& tone_buffer) {} + // DEPRECATED: Older API without tone buffer. + // TODO(bugs.webrtc.org/9725): Remove old API and default implementation + // when old callers are gone. + virtual void OnToneChange(const std::string& tone) {} + + protected: + virtual ~DtmfSenderObserverInterface() = default; +}; + +// The interface of native implementation of the RTCDTMFSender defined by the +// WebRTC W3C Editor's Draft. +// See: https://www.w3.org/TR/webrtc/#peer-to-peer-dtmf +class DtmfSenderInterface : public rtc::RefCountInterface { + public: + // Provides the spec compliant default 2 second delay for the ',' character. + static const int kDtmfDefaultCommaDelayMs = 2000; + + // Used to receive events from the DTMF sender. Only one observer can be + // registered at a time. UnregisterObserver should be called before the + // observer object is destroyed. + virtual void RegisterObserver(DtmfSenderObserverInterface* observer) = 0; + virtual void UnregisterObserver() = 0; + + // Returns true if this DtmfSender is capable of sending DTMF. Otherwise + // returns false. To be able to send DTMF, the associated RtpSender must be + // able to send packets, and a "telephone-event" codec must be negotiated. + virtual bool CanInsertDtmf() = 0; + + // Queues a task that sends the DTMF `tones`. The `tones` parameter is treated + // as a series of characters. The characters 0 through 9, A through D, #, and + // * generate the associated DTMF tones. The characters a to d are equivalent + // to A to D. The character ',' indicates a delay of 2 seconds before + // processing the next character in the tones parameter. + // + // Unrecognized characters are ignored. + // + // The `duration` parameter indicates the duration in ms to use for each + // character passed in the `tones` parameter. The duration cannot be more + // than 6000 or less than 70. + // + // The `inter_tone_gap` parameter indicates the gap between tones in ms. The + // `inter_tone_gap` must be at least 50 ms but should be as short as + // possible. + // + // The `comma_delay` parameter indicates the delay after the ',' + // character. InsertDtmf specifies `comma_delay` as an argument + // with a default value of 2 seconds as per the WebRTC spec. This parameter + // allows users to comply with legacy WebRTC clients. The `comma_delay` + // must be at least 50 ms. + // + // If InsertDtmf is called on the same object while an existing task for this + // object to generate DTMF is still running, the previous task is canceled. + // Returns true on success and false on failure. + virtual bool InsertDtmf(const std::string& tones, + int duration, + int inter_tone_gap) { + return InsertDtmf(tones, duration, inter_tone_gap, + kDtmfDefaultCommaDelayMs); + } + virtual bool InsertDtmf(const std::string& tones, + int duration, + int inter_tone_gap, + int comma_delay) { + // TODO(bugs.webrtc.org/165700): Remove once downstream implementations + // override this signature rather than the 3-parameter one. + return InsertDtmf(tones, duration, inter_tone_gap); + } + + // Returns the tones remaining to be played out. + virtual std::string tones() const = 0; + + // Returns the current tone duration value in ms. + // This value will be the value last set via the InsertDtmf() method, or the + // default value of 100 ms if InsertDtmf() was never called. + virtual int duration() const = 0; + + // Returns the current value of the between-tone gap in ms. + // This value will be the value last set via the InsertDtmf() method, or the + // default value of 50 ms if InsertDtmf() was never called. + virtual int inter_tone_gap() const = 0; + + // Returns the current value of the "," character delay in ms. + // This value will be the value last set via the InsertDtmf() method, or the + // default value of 2000 ms if InsertDtmf() was never called. + virtual int comma_delay() const { return kDtmfDefaultCommaDelayMs; } + + protected: + ~DtmfSenderInterface() override = default; +}; + +} // namespace webrtc + +#endif // API_DTMF_SENDER_INTERFACE_H_ |