5804936052
And use a std::min. Post-commit fixes after: https://review.webrtc.org/30779004/ TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/25059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d
111 lines
3.5 KiB
C++
111 lines
3.5 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/audio_converter.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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namespace webrtc {
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namespace {
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void DownmixToMono(const float* const* src,
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int src_channels,
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int frames,
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float* dst) {
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DCHECK_GT(src_channels, 0);
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for (int i = 0; i < frames; ++i) {
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float sum = 0;
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for (int j = 0; j < src_channels; ++j)
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sum += src[j][i];
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dst[i] = sum / src_channels;
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}
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}
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void UpmixFromMono(const float* src,
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int dst_channels,
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int frames,
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float* const* dst) {
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DCHECK_GT(dst_channels, 0);
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for (int i = 0; i < frames; ++i) {
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float value = src[i];
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for (int j = 0; j < dst_channels; ++j)
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dst[j][i] = value;
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}
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}
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} // namespace
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AudioConverter::AudioConverter(int src_channels, int src_frames,
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int dst_channels, int dst_frames)
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: src_channels_(src_channels),
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src_frames_(src_frames),
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dst_channels_(dst_channels),
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dst_frames_(dst_frames) {
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CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
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const int resample_channels = std::min(src_channels, dst_channels);
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// Prepare buffers as needed for intermediate stages.
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if (dst_channels < src_channels)
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downmix_buffer_.reset(new ChannelBuffer<float>(src_frames,
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resample_channels));
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if (src_frames != dst_frames) {
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resamplers_.reserve(resample_channels);
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for (int i = 0; i < resample_channels; ++i)
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resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
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}
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}
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void AudioConverter::Convert(const float* const* src,
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int src_channels,
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int src_frames,
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int dst_channels,
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int dst_frames,
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float* const* dst) {
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DCHECK_EQ(src_channels_, src_channels);
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DCHECK_EQ(src_frames_, src_frames);
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DCHECK_EQ(dst_channels_, dst_channels);
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DCHECK_EQ(dst_frames_, dst_frames);;
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if (src_channels == dst_channels && src_frames == dst_frames) {
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// Shortcut copy.
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if (src != dst) {
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for (int i = 0; i < src_channels; ++i)
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memcpy(dst[i], src[i], dst_frames * sizeof(*dst[i]));
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}
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return;
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}
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const float* const* src_ptr = src;
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if (dst_channels < src_channels) {
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float* const* dst_ptr = dst;
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if (src_frames != dst_frames) {
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// Downmix to a buffer for subsequent resampling.
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DCHECK_EQ(downmix_buffer_->num_channels(), dst_channels);
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DCHECK_EQ(downmix_buffer_->samples_per_channel(), src_frames);
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dst_ptr = downmix_buffer_->channels();
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}
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DownmixToMono(src, src_channels, src_frames, dst_ptr[0]);
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src_ptr = dst_ptr;
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}
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if (src_frames != dst_frames) {
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for (size_t i = 0; i < resamplers_.size(); ++i)
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resamplers_[i]->Resample(src_ptr[i], src_frames, dst[i], dst_frames);
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src_ptr = dst;
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}
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if (dst_channels > src_channels)
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UpmixFromMono(src_ptr[0], dst_channels, dst_frames, dst);
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}
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} // namespace webrtc
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