diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc | 59 |
1 files changed, 59 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc new file mode 100644 index 0000000000..5633f11b86 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc @@ -0,0 +1,59 @@ +/* + * 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) { + RTC_DCHECK(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_; + RTC_DCHECK_GT(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 |