Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/tools/rtp_generator.h

@@ -0,0 +1,83 @@
+/*
+ *  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_TOOLS_RTP_GENERATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for generating RTP headers.
+class RtpGenerator {
+ public:
+  RtpGenerator(int samples_per_ms,
+               uint16_t start_seq_number = 0,
+               uint32_t start_timestamp = 0,
+               uint32_t start_send_time_ms = 0,
+               uint32_t ssrc = 0x12345678)
+      : seq_number_(start_seq_number),
+        timestamp_(start_timestamp),
+        next_send_time_ms_(start_send_time_ms),
+        ssrc_(ssrc),
+        samples_per_ms_(samples_per_ms),
+        drift_factor_(0.0) {
+  }
+
+  virtual ~RtpGenerator() {}
+
+  // Writes the next RTP header to |rtp_header|, which will be of type
+  // |payload_type|. Returns the send time for this packet (in ms). The value of
+  // |payload_length_samples| determines the send time for the next packet.
+  virtual uint32_t GetRtpHeader(uint8_t payload_type,
+                                size_t payload_length_samples,
+                                RTPHeader* rtp_header);
+
+  void set_drift_factor(double factor);
+
+ protected:
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  uint32_t next_send_time_ms_;
+  const uint32_t ssrc_;
+  const int samples_per_ms_;
+  double drift_factor_;
+
+ private:
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
+};
+
+class TimestampJumpRtpGenerator : public RtpGenerator {
+ public:
+  TimestampJumpRtpGenerator(int samples_per_ms,
+                            uint16_t start_seq_number,
+                            uint32_t start_timestamp,
+                            uint32_t jump_from_timestamp,
+                            uint32_t jump_to_timestamp)
+      : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
+        jump_from_timestamp_(jump_from_timestamp),
+        jump_to_timestamp_(jump_to_timestamp) {}
+
+  uint32_t GetRtpHeader(uint8_t payload_type,
+                        size_t payload_length_samples,
+                        RTPHeader* rtp_header) override;
+
+ private:
+  uint32_t jump_from_timestamp_;
+  uint32_t jump_to_timestamp_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_