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webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

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Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}
This commit is contained in:
kwiberg
2016-09-14 05:23:22 -07:00
committed by Commit bot
parent 3a7f35b1c4
commit 9e2be5f292
27 changed files with 159 additions and 149 deletions
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33
webrtc/modules/audio_processing/audio_buffer.cc
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@ -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();