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. |