diff options
Diffstat (limited to 'webrtc/voice_engine/utility.cc')
-rw-r--r-- | webrtc/voice_engine/utility.cc | 27 |
1 files changed, 15 insertions, 12 deletions
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc index 7bc7e0e963..605e55369e 100644 --- a/webrtc/voice_engine/utility.cc +++ b/webrtc/voice_engine/utility.cc @@ -10,12 +10,12 @@ #include "webrtc/voice_engine/utility.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" #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" -#include "webrtc/system_wrappers/include/logging.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/voice_engine/voice_engine_defines.h" namespace webrtc { @@ -34,12 +34,12 @@ void RemixAndResample(const AudioFrame& src_frame, void RemixAndResample(const int16_t* src_data, size_t samples_per_channel, - int num_channels, + size_t num_channels, int sample_rate_hz, PushResampler<int16_t>* resampler, AudioFrame* dst_frame) { const int16_t* audio_ptr = src_data; - int audio_ptr_num_channels = num_channels; + size_t audio_ptr_num_channels = num_channels; int16_t mono_audio[AudioFrame::kMaxDataSizeSamples]; // Downmix before resampling. @@ -52,8 +52,10 @@ void RemixAndResample(const int16_t* src_data, if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, audio_ptr_num_channels) == -1) { - LOG_FERR3(LS_ERROR, InitializeIfNeeded, sample_rate_hz, - dst_frame->sample_rate_hz_, audio_ptr_num_channels); + 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); } @@ -61,11 +63,12 @@ void RemixAndResample(const int16_t* src_data, int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_, AudioFrame::kMaxDataSizeSamples); if (out_length == -1) { - LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_); + LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr + << ", src_length = " << src_length + << ", dst_frame->data_ = " << dst_frame->data_; assert(false); } - dst_frame->samples_per_channel_ = - static_cast<size_t>(out_length / audio_ptr_num_channels); + dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; // Upmix after resampling. if (num_channels == 1 && dst_frame->num_channels_ == 2) { @@ -77,9 +80,9 @@ void RemixAndResample(const int16_t* src_data, } void MixWithSat(int16_t target[], - int target_channel, + size_t target_channel, const int16_t source[], - int source_channel, + size_t source_channel, size_t source_len) { assert(target_channel == 1 || target_channel == 2); assert(source_channel == 1 || source_channel == 2); |