/* * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #include "common_audio/resampler/include/push_resampler.h" #include #include #include #include "common_audio/include/audio_util.h" #include "common_audio/resampler/push_sinc_resampler.h" #include "rtc_base/checks.h" namespace webrtc { template PushResampler::PushResampler() : src_sample_rate_hz_(0), dst_sample_rate_hz_(0), num_channels_(0) {} template PushResampler::~PushResampler() {} template int PushResampler::InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, size_t num_channels) { // These checks used to be factored out of this template function due to // Windows debug build issues with clang. http://crbug.com/615050 RTC_DCHECK_GT(src_sample_rate_hz, 0); RTC_DCHECK_GT(dst_sample_rate_hz, 0); RTC_DCHECK_GT(num_channels, 0); if (src_sample_rate_hz == src_sample_rate_hz_ && dst_sample_rate_hz == dst_sample_rate_hz_ && 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) { return -1; } src_sample_rate_hz_ = src_sample_rate_hz; dst_sample_rate_hz_ = dst_sample_rate_hz; num_channels_ = num_channels; const size_t src_size_10ms_mono = static_cast(src_sample_rate_hz / 100); const size_t dst_size_10ms_mono = static_cast(dst_sample_rate_hz / 100); channel_resamplers_.clear(); for (size_t i = 0; i < num_channels; ++i) { channel_resamplers_.push_back(ChannelResampler()); auto channel_resampler = channel_resamplers_.rbegin(); channel_resampler->resampler = std::make_unique( src_size_10ms_mono, dst_size_10ms_mono); channel_resampler->source.resize(src_size_10ms_mono); channel_resampler->destination.resize(dst_size_10ms_mono); } channel_data_array_.resize(num_channels_); return 0; } template int PushResampler::Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity) { // These checks used to be factored out of this template function due to // Windows debug build issues with clang. http://crbug.com/615050 const size_t src_size_10ms = (src_sample_rate_hz_ / 100) * num_channels_; const size_t dst_size_10ms = (dst_sample_rate_hz_ / 100) * num_channels_; RTC_DCHECK_EQ(src_length, src_size_10ms); RTC_DCHECK_GE(dst_capacity, dst_size_10ms); if (src_sample_rate_hz_ == dst_sample_rate_hz_) { // The old resampler provides this memcpy facility in the case of matching // sample rates, so reproduce it here for the sinc resampler. memcpy(dst, src, src_length * sizeof(T)); return static_cast(src_length); } const size_t src_length_mono = src_length / num_channels_; const size_t dst_capacity_mono = dst_capacity / num_channels_; for (size_t ch = 0; ch < num_channels_; ++ch) { channel_data_array_[ch] = channel_resamplers_[ch].source.data(); } Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data()); size_t dst_length_mono = 0; for (auto& resampler : channel_resamplers_) { dst_length_mono = resampler.resampler->Resample( resampler.source.data(), src_length_mono, resampler.destination.data(), dst_capacity_mono); } for (size_t ch = 0; ch < num_channels_; ++ch) { channel_data_array_[ch] = channel_resamplers_[ch].destination.data(); } Interleave(channel_data_array_.data(), dst_length_mono, num_channels_, dst); return static_cast(dst_length_mono * num_channels_); } // Explictly generate required instantiations. template class PushResampler; template class PushResampler; } // namespace webrtc