commit 25ab3228f3e473f2226f219531ec617d2daa175e
author Mirko Bonadei <mbonadei@webrtc.org> Thu Jul 08 20:08:20 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Fri Jul 09 07:49:43 2021 +0000
tree 4c50ad84da49e0e657c26e2fe21603dc6fa78f6b
parent 9b5d570ae00b44de9bbb7e57b7c845d4f3f83e2b

Replace assert() with RTC_DCHECK().

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}

modules/audio_coding/acm2/acm_send_test.cc

diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 6f395d6..cda668d 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -51,8 +51,8 @@
   input_frame_.sample_rate_hz_ = source_rate_hz_;
   input_frame_.num_channels_ = 1;
   input_frame_.samples_per_channel_ = input_block_size_samples_;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
   acm_->RegisterTransportCallback(this);
 }
 
@@ -81,8 +81,8 @@
       factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
   codec_registered_ = true;
   input_frame_.num_channels_ = num_channels;
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
   return codec_registered_;
 }
 
@@ -90,13 +90,13 @@
     std::unique_ptr<AudioEncoder> external_speech_encoder) {
   input_frame_.num_channels_ = external_speech_encoder->NumChannels();
   acm_->SetEncoder(std::move(external_speech_encoder));
-  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
-         AudioFrame::kMaxDataSizeSamples);
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
   codec_registered_ = true;
 }
 
 std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
-  assert(codec_registered_);
+  RTC_DCHECK(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
     // This payload type should be filtered out. Since the payload type is the
     // same throughout the whole test run, no packet at all will be delivered.
@@ -133,7 +133,7 @@
   payload_type_ = payload_type;
   timestamp_ = timestamp;
   last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
-  assert(last_payload_vec_.size() == payload_len_bytes);
+  RTC_DCHECK_EQ(last_payload_vec_.size(), payload_len_bytes);
   data_to_send_ = true;
   return 0;
 }