Add ctors to ChannelBuffer to enable copying on construction.
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.
R=claguna@google.com
Review URL: https://webrtc-codereview.appspot.com/24469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/include/audio_util.h b/webrtc/common_audio/include/audio_util.h
index 9972a0e..0ce034b 100644
--- a/webrtc/common_audio/include/audio_util.h
+++ b/webrtc/common_audio/include/audio_util.h
@@ -62,7 +62,7 @@
// per buffer).
template <typename T>
void Deinterleave(const T* interleaved, int samples_per_channel,
- int num_channels, T** deinterleaved) {
+ int num_channels, T* const* deinterleaved) {
for (int i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
int interleaved_idx = i;
diff --git a/webrtc/modules/audio_processing/common.h b/webrtc/modules/audio_processing/common.h
index 98e36cb..c1bae89 100644
--- a/webrtc/modules/audio_processing/common.h
+++ b/webrtc/modules/audio_processing/common.h
@@ -14,6 +14,7 @@
#include <assert.h>
#include <string.h>
+#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -42,37 +43,66 @@
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];
+ SetChannelPtrs();
}
+
+ ChannelBuffer(const T* data, 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) {
+ SetChannelPtrs();
+ memcpy(data_.get(), data, length() * sizeof(T));
+ }
+
+ ChannelBuffer(const T* const* channels, 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) {
+ SetChannelPtrs();
+ for (int i = 0; i < num_channels_; ++i)
+ CopyFrom(channels[i], i);
+ }
+
~ChannelBuffer() {}
void CopyFrom(const void* channel_ptr, int i) {
- assert(i < num_channels_);
+ DCHECK_LT(i, num_channels_);
memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
}
T* data() { return data_.get(); }
+ const T* data() const { return data_.get(); }
+
const T* channel(int i) const {
- assert(i >= 0 && i < num_channels_);
+ DCHECK_GE(i, 0);
+ DCHECK_LT(i, num_channels_);
return channels_[i];
}
T* channel(int i) {
const ChannelBuffer<T>* t = this;
return const_cast<T*>(t->channel(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_; }
+ T* const* channels() { return channels_.get(); }
+ const T* const* channels() const { return channels_.get(); }
+
+ int samples_per_channel() const { return samples_per_channel_; }
+ int num_channels() const { return num_channels_; }
+ int length() const { return samples_per_channel_ * num_channels_; }
private:
+ void SetChannelPtrs() {
+ for (int i = 0; i < num_channels_; ++i)
+ channels_[i] = &data_[i * samples_per_channel_];
+ }
+
scoped_ptr<T[]> data_;
scoped_ptr<T*[]> channels_;
- int samples_per_channel_;
- int num_channels_;
+ const int samples_per_channel_;
+ const int num_channels_;
};
} // namespace webrtc
