Index: webrtc/modules/audio_processing/audio_buffer.cc |
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc |
index 13ece67f254f9fcdc5115a58db28e9de411954f5..f5b9016a675759824e840ff52167068a388f965c 100644 |
--- a/webrtc/modules/audio_processing/audio_buffer.cc |
+++ b/webrtc/modules/audio_processing/audio_buffer.cc |
@@ -10,6 +10,7 @@ |
#include "webrtc/modules/audio_processing/audio_buffer.h" |
+#include "webrtc/base/checks.h" |
#include "webrtc/common_audio/include/audio_util.h" |
#include "webrtc/common_audio/resampler/push_sinc_resampler.h" |
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
@@ -25,7 +26,7 @@ const size_t kSamplesPer48kHzChannel = 480; |
int KeyboardChannelIndex(const StreamConfig& stream_config) { |
if (!stream_config.has_keyboard()) { |
- assert(false); |
+ RTC_NOTREACHED(); |
return 0; |
} |
@@ -61,11 +62,12 @@ AudioBuffer::AudioBuffer(size_t input_num_frames, |
activity_(AudioFrame::kVadUnknown), |
keyboard_data_(NULL), |
data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) { |
- assert(input_num_frames_ > 0); |
- assert(proc_num_frames_ > 0); |
- assert(output_num_frames_ > 0); |
- assert(num_input_channels_ > 0); |
- assert(num_proc_channels_ > 0 && num_proc_channels_ <= num_input_channels_); |
+ RTC_DCHECK_GT(input_num_frames_, 0u); |
+ RTC_DCHECK_GT(proc_num_frames_, 0u); |
+ RTC_DCHECK_GT(output_num_frames_, 0u); |
+ RTC_DCHECK_GT(num_input_channels_, 0u); |
+ RTC_DCHECK_GT(num_proc_channels_, 0u); |
+ RTC_DCHECK_LE(num_proc_channels_, num_input_channels_); |
if (input_num_frames_ != proc_num_frames_ || |
output_num_frames_ != proc_num_frames_) { |
@@ -102,8 +104,8 @@ AudioBuffer::~AudioBuffer() {} |
void AudioBuffer::CopyFrom(const float* const* data, |
const StreamConfig& stream_config) { |
- assert(stream_config.num_frames() == input_num_frames_); |
- assert(stream_config.num_channels() == num_input_channels_); |
+ RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_); |
+ RTC_DCHECK_EQ(stream_config.num_channels(), num_input_channels_); |
InitForNewData(); |
// Initialized lazily because there's a different condition in |
// DeinterleaveFrom. |
@@ -147,8 +149,9 @@ void AudioBuffer::CopyFrom(const float* const* data, |
void AudioBuffer::CopyTo(const StreamConfig& stream_config, |
float* const* data) { |
- assert(stream_config.num_frames() == output_num_frames_); |
- assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1); |
+ RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_); |
+ RTC_DCHECK(stream_config.num_channels() == num_channels_ || |
+ num_channels_ == 1); |
// Convert to the float range. |
float* const* data_ptr = data; |
@@ -374,8 +377,8 @@ size_t AudioBuffer::num_bands() const { |
// The resampler is only for supporting 48kHz to 16kHz in the reverse stream. |
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) { |
- assert(frame->num_channels_ == num_input_channels_); |
- assert(frame->samples_per_channel_ == input_num_frames_); |
+ RTC_DCHECK_EQ(frame->num_channels_, num_input_channels_); |
+ RTC_DCHECK_EQ(frame->samples_per_channel_, input_num_frames_); |
InitForNewData(); |
// Initialized lazily because there's a different condition in CopyFrom. |
if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) { |
@@ -395,7 +398,7 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) { |
DownmixInterleavedToMono(frame->data_, input_num_frames_, |
num_input_channels_, deinterleaved[0]); |
} else { |
- assert(num_proc_channels_ == num_input_channels_); |
+ RTC_DCHECK_EQ(num_proc_channels_, num_input_channels_); |
Deinterleave(frame->data_, |
input_num_frames_, |
num_proc_channels_, |
@@ -419,8 +422,8 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) { |
return; |
} |
- assert(frame->num_channels_ == num_channels_ || num_channels_ == 1); |
- assert(frame->samples_per_channel_ == output_num_frames_); |
+ RTC_DCHECK(frame->num_channels_ == num_channels_ || num_channels_ == 1); |
+ RTC_DCHECK_EQ(frame->samples_per_channel_, output_num_frames_); |
// Resample if necessary. |
IFChannelBuffer* data_ptr = data_.get(); |