Add a simple AudioConverter class.
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.
The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.
BUG=chromium:405270
R=aluebs@webrtc.org, mgraczyk@chromium.org
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/30779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
new file mode 100644
index 0000000..9e18033
--- /dev/null
+++ b/webrtc/common_audio/audio_converter.cc
@@ -0,0 +1,104 @@
+/*
+ * 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) {
+ CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
+ const int resample_channels = src_channels < dst_channels ? 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(dst_channels == src_channels || dst_channels == 1 ||
+ src_channels == 1);
+ 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
