Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_coding/neteq/dtmf_buffer.h

@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+
+#include <list>
+#include <string>  // size_t
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct DtmfEvent {
+  uint32_t timestamp;
+  int event_no;
+  int volume;
+  int duration;
+  bool end_bit;
+
+  // Constructors
+  DtmfEvent()
+      : timestamp(0),
+        event_no(0),
+        volume(0),
+        duration(0),
+        end_bit(false) {
+  }
+  DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
+      : timestamp(ts),
+        event_no(ev),
+        volume(vol),
+        duration(dur),
+        end_bit(end) {
+  }
+};
+
+// This is the buffer holding DTMF events while waiting for them to be played.
+class DtmfBuffer {
+ public:
+  enum BufferReturnCodes {
+    kOK = 0,
+    kInvalidPointer,
+    kPayloadTooShort,
+    kInvalidEventParameters,
+    kInvalidSampleRate
+  };
+
+  // Set up the buffer for use at sample rate |fs_hz|.
+  explicit DtmfBuffer(int fs_hz) {
+    SetSampleRate(fs_hz);
+  }
+
+  virtual ~DtmfBuffer() {}
+
+  // Flushes the buffer.
+  virtual void Flush() { buffer_.clear(); }
+
+  // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
+  // and write the parsed information into the struct |event|. Input variable
+  // |rtp_timestamp| is simply copied into the struct.
+  static int ParseEvent(uint32_t rtp_timestamp,
+                        const uint8_t* payload,
+                        int payload_length_bytes,
+                        DtmfEvent* event);
+
+  // Inserts |event| into the buffer. The method looks for a matching event and
+  // merges the two if a match is found.
+  virtual int InsertEvent(const DtmfEvent& event);
+
+  // Checks if a DTMF event should be played at time |current_timestamp|. If so,
+  // the method returns true; otherwise false. The parameters of the event to
+  // play will be written to |event|.
+  virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
+
+  // Number of events in the buffer.
+  virtual size_t Length() const { return buffer_.size(); }
+
+  virtual bool Empty() const { return buffer_.empty(); }
+
+  // Set a new sample rate.
+  virtual int SetSampleRate(int fs_hz);
+
+ private:
+  typedef std::list<DtmfEvent> DtmfList;
+
+  int max_extrapolation_samples_;
+  int frame_len_samples_;  // TODO(hlundin): Remove this later.
+
+  // Compares two events and returns true if they are the same.
+  static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
+
+  // Merges |event| to the event pointed out by |it|. The method checks that
+  // the two events are the same (using the SameEvent method), and merges them
+  // if that was the case, returning true. If the events are not the same, false
+  // is returned.
+  bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
+
+  // Method used by the sort algorithm to rank events in the buffer.
+  static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
+
+  DtmfList buffer_;
+
+  DISALLOW_COPY_AND_ASSIGN(DtmfBuffer);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_