mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
25ab3228f3
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
61 lines
2 KiB
C++
61 lines
2 KiB
C++
/*
|
|
* 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"
|
|
|
|
#include <assert.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
|