Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 104 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
new file mode 100644
index 0000000000..62b751525c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
@@ -0,0 +1,104 @@
+/*
+ *  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 MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+
+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);
+
+  virtual ~DtmfBuffer();
+
+  DtmfBuffer(const DtmfBuffer&) = delete;
+  DtmfBuffer& operator=(const DtmfBuffer&) = delete;
+
+  // Flushes the buffer.
+  virtual void Flush();
+
+  // 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,
+                        size_t 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;
+
+  virtual bool Empty() const;
+
+  // 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_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_