| /* |
| * 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 |