| Index: webrtc/common_audio/resampler/push_resampler.cc
|
| diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc
|
| index f654e9a3978802df327ddb5fd99eb2a431b2098f..06fdfb814d45bc543f915489d1daf662650a8bbb 100644
|
| --- a/webrtc/common_audio/resampler/push_resampler.cc
|
| +++ b/webrtc/common_audio/resampler/push_resampler.cc
|
| @@ -12,6 +12,7 @@
|
|
|
| #include <string.h>
|
|
|
| +#include "webrtc/base/checks.h"
|
| #include "webrtc/common_audio/include/audio_util.h"
|
| #include "webrtc/common_audio/resampler/include/resampler.h"
|
| #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
|
| @@ -33,15 +34,22 @@
|
| int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
|
| int dst_sample_rate_hz,
|
| size_t num_channels) {
|
| + RTC_DCHECK_GT(src_sample_rate_hz, 0);
|
| + RTC_DCHECK_GT(dst_sample_rate_hz, 0);
|
| + RTC_DCHECK_GT(num_channels, 0u);
|
| + RTC_DCHECK_LE(num_channels, 2u);
|
| +
|
| if (src_sample_rate_hz == src_sample_rate_hz_ &&
|
| dst_sample_rate_hz == dst_sample_rate_hz_ &&
|
| - num_channels == num_channels_)
|
| + num_channels == num_channels_) {
|
| // No-op if settings haven't changed.
|
| return 0;
|
| + }
|
|
|
| - if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
|
| - num_channels <= 0 || num_channels > 2)
|
| + if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 || num_channels <= 0 ||
|
| + num_channels > 2) {
|
| return -1;
|
| + }
|
|
|
| src_sample_rate_hz_ = src_sample_rate_hz;
|
| dst_sample_rate_hz_ = dst_sample_rate_hz;
|
| @@ -70,6 +78,8 @@
|
| size_t dst_capacity) {
|
| const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
|
| const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
|
| + RTC_CHECK_EQ(src_length, src_size_10ms);
|
| + RTC_CHECK_GE(dst_capacity, dst_size_10ms);
|
| if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
|
| return -1;
|
|
|
|
|