| /* |
| * Copyright (c) 2014 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 "webrtc/base/checks.h" |
| #include "webrtc/common_audio/audio_converter.h" |
| #include "webrtc/common_audio/resampler/push_sinc_resampler.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| void DownmixToMono(const float* const* src, |
| int src_channels, |
| int frames, |
| float* dst) { |
| DCHECK_GT(src_channels, 0); |
| for (int i = 0; i < frames; ++i) { |
| float sum = 0; |
| for (int j = 0; j < src_channels; ++j) |
| sum += src[j][i]; |
| dst[i] = sum / src_channels; |
| } |
| } |
| |
| void UpmixFromMono(const float* src, |
| int dst_channels, |
| int frames, |
| float* const* dst) { |
| DCHECK_GT(dst_channels, 0); |
| for (int i = 0; i < frames; ++i) { |
| float value = src[i]; |
| for (int j = 0; j < dst_channels; ++j) |
| dst[j][i] = value; |
| } |
| } |
| |
| } // namespace |
| |
| AudioConverter::AudioConverter(int src_channels, int src_frames, |
| int dst_channels, int dst_frames) |
| : src_channels_(src_channels), |
| src_frames_(src_frames), |
| dst_channels_(dst_channels), |
| dst_frames_(dst_frames) { |
| CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1); |
| const int resample_channels = std::min(src_channels, dst_channels); |
| |
| // Prepare buffers as needed for intermediate stages. |
| if (dst_channels < src_channels) |
| downmix_buffer_.reset(new ChannelBuffer<float>(src_frames, |
| resample_channels)); |
| |
| if (src_frames != dst_frames) { |
| resamplers_.reserve(resample_channels); |
| for (int i = 0; i < resample_channels; ++i) |
| resamplers_.push_back(new PushSincResampler(src_frames, dst_frames)); |
| } |
| } |
| |
| void AudioConverter::Convert(const float* const* src, |
| int src_channels, |
| int src_frames, |
| int dst_channels, |
| int dst_frames, |
| float* const* dst) { |
| DCHECK_EQ(src_channels_, src_channels); |
| DCHECK_EQ(src_frames_, src_frames); |
| DCHECK_EQ(dst_channels_, dst_channels); |
| DCHECK_EQ(dst_frames_, dst_frames);; |
| |
| if (src_channels == dst_channels && src_frames == dst_frames) { |
| // Shortcut copy. |
| if (src != dst) { |
| for (int i = 0; i < src_channels; ++i) |
| memcpy(dst[i], src[i], dst_frames * sizeof(*dst[i])); |
| } |
| return; |
| } |
| |
| const float* const* src_ptr = src; |
| if (dst_channels < src_channels) { |
| float* const* dst_ptr = dst; |
| if (src_frames != dst_frames) { |
| // Downmix to a buffer for subsequent resampling. |
| DCHECK_EQ(downmix_buffer_->num_channels(), dst_channels); |
| DCHECK_EQ(downmix_buffer_->samples_per_channel(), src_frames); |
| dst_ptr = downmix_buffer_->channels(); |
| } |
| |
| DownmixToMono(src, src_channels, src_frames, dst_ptr[0]); |
| src_ptr = dst_ptr; |
| } |
| |
| if (src_frames != dst_frames) { |
| for (size_t i = 0; i < resamplers_.size(); ++i) |
| resamplers_[i]->Resample(src_ptr[i], src_frames, dst[i], dst_frames); |
| src_ptr = dst; |
| } |
| |
| if (dst_channels > src_channels) |
| UpmixFromMono(src_ptr[0], dst_channels, dst_frames, dst); |
| } |
| |
| } // namespace webrtc |