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Fix constness of AudioBuffer accessors.

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Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-04-30 16:44:13 +00:00
parent 740e6b339a
commit 65f933899b
12 changed files with 64 additions and 46 deletions
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28
webrtc/modules/audio_processing/level_estimator_impl.cc
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@ -20,7 +20,15 @@
namespace webrtc {
namespace {
const double kMaxSquaredLevel = 32768.0 * 32768.0;
const float kMaxSquaredLevel = 32768.0 * 32768.0;
float SumSquare(const int16_t* data, int length) {
float sum_square = 0.f;
for (int i = 0; i < length; ++i) {
sum_square += data[i] * data[i];
}
return sum_square;
}
class Level {
public:
@ -36,7 +44,7 @@ class Level {
sample_count_ = 0;
}
void Process(int16_t* data, int length) {
void Process(const int16_t* data, int length) {
assert(data != NULL);
assert(length > 0);
sum_square_ += SumSquare(data, length);
@ -55,7 +63,7 @@ class Level {
}
// Normalize by the max level.
double rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
float rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
// 20log_10(x^0.5) = 10log_10(x)
rms = 10 * log10(rms);
if (rms > 0)
@ -69,18 +77,10 @@ class Level {
}
private:
static double SumSquare(int16_t* data, int length) {
double sum_square = 0.0;
for (int i = 0; i < length; ++i) {
double data_d = static_cast<double>(data[i]);
sum_square += data_d * data_d;
}
return sum_square;
}
double sum_square_;
float sum_square_;
int sample_count_;
};
} // namespace
LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm,
@ -102,7 +102,7 @@ int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
return apm_->kNoError;
}
int16_t* mixed_data = audio->data(0);
const int16_t* mixed_data = audio->data(0);
if (audio->num_channels() > 1) {
audio->CopyAndMix(1);
mixed_data = audio->mixed_data(0);