Adding upstream version 86.0.1.upstream/86.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 97 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/api/dtmfsenderinterface.h b/third_party/libwebrtc/webrtc/api/dtmfsenderinterface.h
new file mode 100644
index 0000000000..8f0ab71206
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/api/dtmfsenderinterface.h
@@ -0,0 +1,97 @@
+/*
+ *  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_DTMFSENDERINTERFACE_H_
+#define API_DTMFSENDERINTERFACE_H_
+
+#include <string>
+
+#include "api/mediastreaminterface.h"
+#include "rtc_base/refcount.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.
+  virtual void OnToneChange(const std::string& tone) = 0;
+
+ protected:
+  virtual ~DtmfSenderObserverInterface() {}
+};
+
+// 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:
+  // 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.
+  //
+  // 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) = 0;
+
+  // Returns the track given as argument to the constructor. Only exists for
+  // backwards compatibilty; now that DtmfSenders are tied to RtpSenders, it's
+  // no longer relevant.
+  virtual const AudioTrackInterface* track() const = 0;
+
+  // 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;
+
+ protected:
+  virtual ~DtmfSenderInterface() {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_DTMFSENDERINTERFACE_H_