Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
new file mode 100644
index 0000000..e4ac6ee
--- /dev/null
+++ b/webrtc/modules/audio_processing/common.h
@@ -0,0 +1,75 @@
+/*
+ * 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
+
+#include <string.h>
+
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+ switch (layout) {
+ case AudioProcessing::kMono:
+ case AudioProcessing::kMonoAndKeyboard:
+ return 1;
+ case AudioProcessing::kStereo:
+ case AudioProcessing::kStereoAndKeyboard:
+ return 2;
+ }
+ assert(false);
+ return -1;
+}
+
+// Helper to encapsulate a contiguous data buffer with access to a pointer
+// array of the deinterleaved channels.
+template <typename T>
+class ChannelBuffer {
+ public:
+ ChannelBuffer(int samples_per_channel, int num_channels)
+ : data_(new T[samples_per_channel * num_channels]),
+ channels_(new T*[num_channels]),
+ samples_per_channel_(samples_per_channel),
+ num_channels_(num_channels) {
+ memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
+ for (int i = 0; i < num_channels; ++i)
+ channels_[i] = &data_[i * samples_per_channel];
+ }
+ ~ChannelBuffer() {}
+
+ void CopyFrom(const void* channel_ptr, int i) {
+ assert(i < num_channels_);
+ memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
+ }
+
+ T* data() { return data_.get(); }
+ T* channel(int i) {
+ assert(i < num_channels_);
+ return channels_[i];
+ }
+ T** channels() { return channels_.get(); }
+
+ int samples_per_channel() { return samples_per_channel_; }
+ int num_channels() { return num_channels_; }
+ int length() { return samples_per_channel_ * num_channels_; }
+
+ private:
+ scoped_ptr<T[]> data_;
+ scoped_ptr<T*[]> channels_;
+ int samples_per_channel_;
+ int num_channels_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_