1/*
2 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#include <assert.h>
12
13#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
14
15namespace webrtc {
16namespace test {
17
18uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
19                                    size_t payload_length_samples,
20                                    WebRtcRTPHeader* rtp_header) {
21  assert(rtp_header);
22  if (!rtp_header) {
23    return 0;
24  }
25  rtp_header->header.sequenceNumber = seq_number_++;
26  rtp_header->header.timestamp = timestamp_;
27  timestamp_ += static_cast<uint32_t>(payload_length_samples);
28  rtp_header->header.payloadType = payload_type;
29  rtp_header->header.markerBit = false;
30  rtp_header->header.ssrc = ssrc_;
31  rtp_header->header.numCSRCs = 0;
32  rtp_header->frameType = kAudioFrameSpeech;
33
34  uint32_t this_send_time = next_send_time_ms_;
35  assert(samples_per_ms_ > 0);
36  next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
37      samples_per_ms_;
38  return this_send_time;
39}
40
41void RtpGenerator::set_drift_factor(double factor) {
42  if (factor > -1.0) {
43    drift_factor_ = factor;
44  }
45}
46
47}  // namespace test
48}  // namespace webrtc
49