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.cc

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include <assert.h>
+
+#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+
+namespace webrtc {
+namespace test {
+
+uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                    size_t payload_length_samples,
+                                    RTPHeader* rtp_header) {
+  assert(rtp_header);
+  if (!rtp_header) {
+    return 0;
+  }
+  rtp_header->sequenceNumber = seq_number_++;
+  rtp_header->timestamp = timestamp_;
+  timestamp_ += static_cast<uint32_t>(payload_length_samples);
+  rtp_header->payloadType = payload_type;
+  rtp_header->markerBit = false;
+  rtp_header->ssrc = ssrc_;
+  rtp_header->numCSRCs = 0;
+
+  uint32_t this_send_time = next_send_time_ms_;
+  assert(samples_per_ms_ > 0);
+  next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
+      samples_per_ms_;
+  return this_send_time;
+}
+
+void RtpGenerator::set_drift_factor(double factor) {
+  if (factor > -1.0) {
+    drift_factor_ = factor;
+  }
+}
+
+uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                                 size_t payload_length_samples,
+                                                 RTPHeader* rtp_header) {
+  uint32_t ret = RtpGenerator::GetRtpHeader(
+      payload_type, payload_length_samples, rtp_header);
+  if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
+          jump_from_timestamp_ &&
+      timestamp_ > jump_from_timestamp_) {
+    // We just moved across the |jump_from_timestamp_| timestamp. Do the jump.
+    timestamp_ = jump_to_timestamp_;
+  }
+  return ret;
+}
+
+}  // namespace test
+}  // namespace webrtc