chromium/third_party/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc

/*
 *  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 "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) { … }

void RtpGenerator::set_drift_factor(double factor) { … }

uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
                                                 size_t payload_length_samples,
                                                 RTPHeader* rtp_header) { … }

}  // namespace test
}  // namespace webrtc