blob: db9988d904d48957b7e492c1ded7499606c4ddab [file] [log] [blame]
/*
* 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;
}
}
uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
WebRtcRTPHeader* 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