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Unified Diff: webrtc/voice_engine/utility.cc

Issue 2007563002: Adding a some checks and switching out a few assert for RTC_[D]CHECK. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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Index: webrtc/voice_engine/utility.cc
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index 605e55369e3d4a31666e17bf69d3b04ccd054630..3a5ea86bec97299b140955372987ccc2dbd1411d 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -10,6 +10,7 @@
#include "webrtc/voice_engine/utility.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@@ -52,21 +53,19 @@ void RemixAndResample(const int16_t* src_data,
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
- LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
- << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
- << dst_frame->sample_rate_hz_
- << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
- assert(false);
+ RTC_CHECK(false) << "InitializeIfNeeded failed: sample_rate_hz = "
hlundin-webrtc 2016/05/24 06:38:03 Consider FATAL().
tommi 2016/05/24 07:49:13 Done.
+ << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
+ << dst_frame->sample_rate_hz_
+ << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
}
const size_t src_length = samples_per_channel * audio_ptr_num_channels;
int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
- LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
- << ", src_length = " << src_length
- << ", dst_frame->data_ = " << dst_frame->data_;
- assert(false);
+ RTC_CHECK(false) << "Resample failed: audio_ptr = " << audio_ptr
hlundin-webrtc 2016/05/24 06:38:03 And here.
tommi 2016/05/24 07:49:13 Done.
+ << ", src_length = " << src_length
+ << ", dst_frame->data_ = " << dst_frame->data_;
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
@@ -84,8 +83,8 @@ void MixWithSat(int16_t target[],
const int16_t source[],
size_t source_channel,
size_t source_len) {
- assert(target_channel == 1 || target_channel == 2);
- assert(source_channel == 1 || source_channel == 2);
+ RTC_DCHECK(target_channel == 1 || target_channel == 2);
hlundin-webrtc 2016/05/24 06:38:03 Consider rewriting these as RTC_DCHECK_GE(target_c
tommi 2016/05/24 07:49:13 Done.
+ RTC_DCHECK(source_channel == 1 || source_channel == 2);
if (target_channel == 2 && source_channel == 1) {
// Convert source from mono to stereo.
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