Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/dtmf_buffer.h b/webrtc/modules/audio_coding/neteq4/dtmf_buffer.h
new file mode 100644
index 0000000..d08b64f
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/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_NETEQ4_DTMF_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DTMF_BUFFER_H_
+
+#include <list>
+#include <string> // size_t
+
+#include "webrtc/system_wrappers/interface/constructor_magic.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_NETEQ4_DTMF_BUFFER_H_