zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Replace assert() with RTC_DCHECK().

Browse Source 
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}
This commit is contained in:
Mirko Bonadei
2021-07-08 20:08:20 +02:00
committed by WebRTC LUCI CQ
parent 9b5d570ae0
commit 25ab3228f3
82 changed files with 407 additions and 392 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_coding/neteq/tools/output_audio_file.h
View File

@ -36,7 +36,7 @@ class OutputAudioFile : public AudioSink {
}
bool WriteArray(const int16_t* audio, size_t num_samples) override {
assert(out_file_);
RTC_DCHECK(out_file_);
return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
}

4
modules/audio_coding/neteq/tools/rtp_analyze.cc
View File

@ -56,7 +56,7 @@ int main(int argc, char* argv[]) {
printf("Input file: %s\n", args[1]);
std::unique_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(args[1]));
assert(file_source.get());
RTC_DCHECK(file_source.get());
// Set RTP extension IDs.
bool print_audio_level = false;
if (absl::GetFlag(FLAGS_audio_level) != -1) {
@ -151,7 +151,7 @@ int main(int argc, char* argv[]) {
packet->ExtractRedHeaders(&red_headers);
while (!red_headers.empty()) {
webrtc::RTPHeader* red = red_headers.front();
assert(red);
RTC_DCHECK(red);
fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
red->timestamp, static_cast<unsigned int>(packet->time_ms()),
red->payloadType);

4
modules/audio_coding/neteq/tools/rtp_generator.cc
View File

@ -18,7 +18,7 @@ namespace test {
uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header) {
assert(rtp_header);
RTC_DCHECK(rtp_header);
if (!rtp_header) {
return 0;
}
@ -31,7 +31,7 @@ uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
rtp_header->numCSRCs = 0;
uint32_t this_send_time = next_send_time_ms_;
assert(samples_per_ms_ > 0);
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;