Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 2338a53..e73acc2 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -119,7 +119,7 @@
rtp_header_,
rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
if (ret_val < 0) {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return -1;
}
rtp_header_.sequenceNumber++;
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index 7c7393d..e307c6c 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -30,7 +30,7 @@
size_t in_length = in_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return -1;
}
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index b742a82..e2081e2 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -344,13 +344,13 @@
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
InputData* input_data) {
if (audio_frame.samples_per_channel_ == 0) {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
return -1;
}
if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
return -1;
}
diff --git a/modules/audio_coding/acm2/call_statistics.cc b/modules/audio_coding/acm2/call_statistics.cc
index 0aad594..9f3bdad 100644
--- a/modules/audio_coding/acm2/call_statistics.cc
+++ b/modules/audio_coding/acm2/call_statistics.cc
@@ -45,7 +45,7 @@
}
case AudioFrame::kUndefined: {
// If the audio is decoded by NetEq, `kUndefined` is not an option.
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
}
}
}
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 6708bc0..87759c3 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -265,7 +265,7 @@
break;
case audio_network_adaptor::config::Controller::kFecControllerRplrBased:
// FecControllerRplrBased has been removed and can't be used anymore.
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
continue;
case audio_network_adaptor::config::Controller::kFrameLengthController:
controller = CreateFrameLengthController(
@@ -293,7 +293,7 @@
encoder_frame_lengths_ms);
break;
default:
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
}
if (controller_config.has_scoring_point()) {
auto& scoring_point = controller_config.scoring_point();
@@ -321,7 +321,7 @@
}
#else
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return nullptr;
#endif // WEBRTC_ENABLE_PROTOBUF
}
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 669cf5e..2616706 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -76,7 +76,7 @@
dump_file_ = FileWrapper(file_handle);
RTC_CHECK(dump_file_.is_open());
#else
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
#endif
}
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 936e224..c5e5fa7 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -26,7 +26,7 @@
absl::optional<float> GetAverage() override { return last_sample_; }
bool SetTimeConstantMs(int time_constant_ms) override {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return false;
}
diff --git a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
index be8d0c6..cafca75 100644
--- a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
+++ b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc
@@ -51,8 +51,8 @@
filename = test::ResourcePath("audio_coding/testfile32kHz", "pcm");
break;
default:
- RTC_NOTREACHED() << "No test file available for " << sample_rate_hz
- << " Hz.";
+ RTC_DCHECK_NOTREACHED()
+ << "No test file available for " << sample_rate_hz << " Hz.";
}
auto pcm_file = std::make_unique<PCMFile>();
pcm_file->ReadStereo(false);
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
index 1d0c7fe..f81aeee 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -88,7 +88,7 @@
samples_per_ms_ = 8;
break;
default:
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
break;
}
}
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 9b1dce8..003d966 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -54,7 +54,7 @@
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes) {
- RTC_NOTREACHED() << "This method must be overridden, or not used.";
+ RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
return -1;
}
};
diff --git a/modules/audio_coding/neteq/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc
index 54ec556..a979f94 100644
--- a/modules/audio_coding/neteq/dsp_helper.cc
+++ b/modules/audio_coding/neteq/dsp_helper.cc
@@ -354,7 +354,7 @@
break;
}
default: {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return -1;
}
}
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.cc b/modules/audio_coding/neteq/dtmf_tone_generator.cc
index 49cbf8f..9061e27 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator.cc
+++ b/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -119,7 +119,7 @@
} else if (fs == 48000) {
fs_index = 3;
} else {
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
fs_index = 1; // Default to 8000 Hz.
}
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index ca5ec22..22cf6a7 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -372,7 +372,7 @@
while (((best_correlation_index + input_length) <
(timestamps_per_call_ + expand_->overlap_length())) ||
((best_correlation_index + input_length) < start_position)) {
- RTC_NOTREACHED(); // Should never happen.
+ RTC_DCHECK_NOTREACHED(); // Should never happen.
best_correlation_index += expand_period; // Jump one lag ahead.
}
return best_correlation_index;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 6107b17..7e3c4ef 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -221,7 +221,7 @@
break;
}
default:
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
}
if (!vad_enabled) {
// Always set kVadUnknown when receive VAD is inactive.
@@ -894,7 +894,7 @@
}
case Operation::kUndefined: {
RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
- RTC_NOTREACHED(); // This should not happen.
+ RTC_DCHECK_NOTREACHED(); // This should not happen.
last_mode_ = Mode::kError;
return kInvalidOperation;
}
@@ -1057,7 +1057,7 @@
// Don't use this packet, discard it.
if (packet_buffer_->DiscardNextPacket(stats_.get()) !=
PacketBuffer::kOK) {
- RTC_NOTREACHED(); // Must be ok by design.
+ RTC_DCHECK_NOTREACHED(); // Must be ok by design.
}
// Check buffer again.
if (!new_codec_) {
@@ -1877,7 +1877,7 @@
// // it must be copied to the speech buffer.
// // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and
// // verify correct operation.
- // RTC_NOTREACHED();
+ // RTC_DCHECK_NOTREACHED();
// // Must generate enough data to replace all of the `sync_buffer_`
// // "future".
// int required_length = sync_buffer_->FutureLength();
@@ -1967,7 +1967,8 @@
next_packet = nullptr;
if (!packet) {
RTC_LOG(LS_ERROR) << "Should always be able to extract a packet here";
- RTC_NOTREACHED(); // Should always be able to extract a packet here.
+ RTC_DCHECK_NOTREACHED(); // Should always be able to extract a packet
+ // here.
return -1;
}
const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
@@ -2001,7 +2002,7 @@
} else if (!has_cng_packet) {
RTC_LOG(LS_WARNING) << "Unknown payload type "
<< static_cast<int>(packet->payload_type);
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
}
if (packet_duration == 0) {
diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc
index 80e1691..7d7cac7 100644
--- a/modules/audio_coding/neteq/sync_buffer.cc
+++ b/modules/audio_coding/neteq/sync_buffer.cc
@@ -28,7 +28,7 @@
next_index_ -= samples_added;
} else {
// This means that we are pushing out future data that was never used.
- // RTC_NOTREACHED();
+ // RTC_DCHECK_NOTREACHED();
// TODO(hlundin): This assert must be disabled to support 60 ms frames.
// This should not happen even for 60 ms frames, but it does. Investigate
// why.
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index 204f169..ee392f2 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -191,7 +191,7 @@
config.sample_rate_hz = 48000;
return config;
default:
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return config;
}
}
@@ -242,7 +242,7 @@
GetCodecConfig<AudioEncoderIsac>(), payload_type);
}
}
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
return nullptr;
}
@@ -259,7 +259,7 @@
case 48000:
return 100;
default:
- RTC_NOTREACHED();
+ RTC_DCHECK_NOTREACHED();
}
return 0;
};