1 files changed, 48 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
new file mode 100644
index 00000000000..17ac209f1d9
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -0,0 +1,48 @@
+/*
+ *  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,
+                                    WebRtcRTPHeader* rtp_header) {
+  assert(rtp_header);
+  if (!rtp_header) {
+    return 0;
+  }
+  rtp_header->header.sequenceNumber = seq_number_++;
+  rtp_header->header.timestamp = timestamp_;
+  timestamp_ += static_cast<uint32_t>(payload_length_samples);
+  rtp_header->header.payloadType = payload_type;
+  rtp_header->header.markerBit = false;
+  rtp_header->header.ssrc = ssrc_;
+  rtp_header->header.numCSRCs = 0;
+  rtp_header->frameType = kAudioFrameSpeech;
+
+  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;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc