Fix constness of AudioBuffer accessors.
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/webrtc@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/common_audio/vad/include/webrtc_vad.h b/common_audio/vad/include/webrtc_vad.h
index 1a6e10a..0538273 100644
--- a/common_audio/vad/include/webrtc_vad.h
+++ b/common_audio/vad/include/webrtc_vad.h
@@ -69,7 +69,7 @@
// returns : 1 - (Active Voice),
// 0 - (Non-active Voice),
// -1 - (Error)
-int WebRtcVad_Process(VadInst* handle, int fs, int16_t* audio_frame,
+int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
int frame_length);
// Checks for valid combinations of |rate| and |frame_length|. We support 10,
diff --git a/common_audio/vad/vad_core.c b/common_audio/vad/vad_core.c
index 80c31f4..98da6ea 100644
--- a/common_audio/vad/vad_core.c
+++ b/common_audio/vad/vad_core.c
@@ -603,7 +603,7 @@
// Calculate VAD decision by first extracting feature values and then calculate
// probability for both speech and background noise.
-int WebRtcVad_CalcVad48khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length) {
int vad;
int i;
@@ -628,7 +628,7 @@
return vad;
}
-int WebRtcVad_CalcVad32khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length)
{
int len, vad;
@@ -650,7 +650,7 @@
return vad;
}
-int WebRtcVad_CalcVad16khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length)
{
int len, vad;
@@ -666,7 +666,7 @@
return vad;
}
-int WebRtcVad_CalcVad8khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length)
{
int16_t feature_vector[kNumChannels], total_power;
diff --git a/common_audio/vad/vad_core.h b/common_audio/vad/vad_core.h
index d6c1da2..202963d 100644
--- a/common_audio/vad/vad_core.h
+++ b/common_audio/vad/vad_core.h
@@ -85,9 +85,9 @@
/****************************************************************************
* WebRtcVad_CalcVad48khz(...)
- * WebRtcVad_CalcVad32khz(...)
- * WebRtcVad_CalcVad16khz(...)
- * WebRtcVad_CalcVad8khz(...)
+ * WebRtcVad_CalcVad32khz(...)
+ * WebRtcVad_CalcVad16khz(...)
+ * WebRtcVad_CalcVad8khz(...)
*
* Calculate probability for active speech and make VAD decision.
*
@@ -103,13 +103,13 @@
* 0 - No active speech
* 1-6 - Active speech
*/
-int WebRtcVad_CalcVad48khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length);
-int WebRtcVad_CalcVad32khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length);
-int WebRtcVad_CalcVad16khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length);
-int WebRtcVad_CalcVad8khz(VadInstT* inst, int16_t* speech_frame,
+int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
int frame_length);
#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/common_audio/vad/vad_sp.c b/common_audio/vad/vad_sp.c
index 41deb3d..e981ad2 100644
--- a/common_audio/vad/vad_sp.c
+++ b/common_audio/vad/vad_sp.c
@@ -24,7 +24,7 @@
// TODO(bjornv): Move this function to vad_filterbank.c.
// Downsampling filter based on splitting filter and allpass functions.
-void WebRtcVad_Downsampling(int16_t* signal_in,
+void WebRtcVad_Downsampling(const int16_t* signal_in,
int16_t* signal_out,
int32_t* filter_state,
int in_length) {
diff --git a/common_audio/vad/vad_sp.h b/common_audio/vad/vad_sp.h
index f84876a..b5e6259 100644
--- a/common_audio/vad/vad_sp.h
+++ b/common_audio/vad/vad_sp.h
@@ -30,7 +30,7 @@
//
// Output:
// - signal_out : Downsampled signal (of length |in_length| / 2).
-void WebRtcVad_Downsampling(int16_t* signal_in,
+void WebRtcVad_Downsampling(const int16_t* signal_in,
int16_t* signal_out,
int32_t* filter_state,
int in_length);
diff --git a/common_audio/vad/webrtc_vad.c b/common_audio/vad/webrtc_vad.c
index 3b31ef5..8a9b931 100644
--- a/common_audio/vad/webrtc_vad.c
+++ b/common_audio/vad/webrtc_vad.c
@@ -68,7 +68,7 @@
return WebRtcVad_set_mode_core(self, mode);
}
-int WebRtcVad_Process(VadInst* handle, int fs, int16_t* audio_frame,
+int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
int frame_length) {
int vad = -1;
VadInstT* self = (VadInstT*) handle;
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index c53d4df..9160f69 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -228,7 +228,7 @@
is_muted_ = false;
}
-int16_t* AudioBuffer::data(int channel) const {
+const int16_t* AudioBuffer::data(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
if (data_ != NULL) {
return data_;
@@ -237,7 +237,12 @@
return channels_->channel(channel);
}
-int16_t* AudioBuffer::low_pass_split_data(int channel) const {
+int16_t* AudioBuffer::data(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<int16_t*>(t->data(channel));
+}
+
+const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
if (split_channels_.get() == NULL) {
return data(channel);
@@ -246,7 +251,12 @@
return split_channels_->low_channel(channel);
}
-int16_t* AudioBuffer::high_pass_split_data(int channel) const {
+int16_t* AudioBuffer::low_pass_split_data(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<int16_t*>(t->low_pass_split_data(channel));
+}
+
+const int16_t* AudioBuffer::high_pass_split_data(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
if (split_channels_.get() == NULL) {
return NULL;
@@ -255,19 +265,24 @@
return split_channels_->high_channel(channel);
}
-int16_t* AudioBuffer::mixed_data(int channel) const {
+int16_t* AudioBuffer::high_pass_split_data(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<int16_t*>(t->high_pass_split_data(channel));
+}
+
+const int16_t* AudioBuffer::mixed_data(int channel) const {
assert(channel >= 0 && channel < num_mixed_channels_);
return mixed_channels_->channel(channel);
}
-int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
+const int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
return mixed_low_pass_channels_->channel(channel);
}
-int16_t* AudioBuffer::low_pass_reference(int channel) const {
+const int16_t* AudioBuffer::low_pass_reference(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
if (!reference_copied_) {
return NULL;
@@ -280,7 +295,7 @@
return keyboard_data_;
}
-SplitFilterStates* AudioBuffer::filter_states(int channel) const {
+SplitFilterStates* AudioBuffer::filter_states(int channel) {
assert(channel >= 0 && channel < num_proc_channels_);
return &filter_states_[channel];
}
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index eaf53eb..79f4689 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -55,15 +55,18 @@
int samples_per_split_channel() const;
int samples_per_keyboard_channel() const;
- int16_t* data(int channel) const;
- int16_t* low_pass_split_data(int channel) const;
- int16_t* high_pass_split_data(int channel) const;
- int16_t* mixed_data(int channel) const;
- int16_t* mixed_low_pass_data(int channel) const;
- int16_t* low_pass_reference(int channel) const;
+ int16_t* data(int channel);
+ const int16_t* data(int channel) const;
+ int16_t* low_pass_split_data(int channel);
+ const int16_t* low_pass_split_data(int channel) const;
+ int16_t* high_pass_split_data(int channel);
+ const int16_t* high_pass_split_data(int channel) const;
+ const int16_t* mixed_data(int channel) const;
+ const int16_t* mixed_low_pass_data(int channel) const;
+ const int16_t* low_pass_reference(int channel) const;
const float* keyboard_data() const;
- SplitFilterStates* filter_states(int channel) const;
+ SplitFilterStates* filter_states(int channel);
void set_activity(AudioFrame::VADActivity activity);
AudioFrame::VADActivity activity() const;
diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc
index 1dce403..a03adc5 100644
--- a/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/modules/audio_processing/echo_control_mobile_impl.cc
@@ -128,7 +128,7 @@
for (int i = 0; i < audio->num_channels(); i++) {
// TODO(ajm): improve how this works, possibly inside AECM.
// This is kind of hacked up.
- int16_t* noisy = audio->low_pass_reference(i);
+ const int16_t* noisy = audio->low_pass_reference(i);
int16_t* clean = audio->low_pass_split_data(i);
if (noisy == NULL) {
noisy = clean;
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index e859044..a67b67e 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -59,7 +59,7 @@
assert(audio->samples_per_split_channel() <= 160);
- int16_t* mixed_data = audio->low_pass_split_data(0);
+ const int16_t* mixed_data = audio->low_pass_split_data(0);
if (audio->num_channels() > 1) {
audio->CopyAndMixLowPass(1);
mixed_data = audio->mixed_low_pass_data(0);
diff --git a/modules/audio_processing/level_estimator_impl.cc b/modules/audio_processing/level_estimator_impl.cc
index c5985ce..a91e963 100644
--- a/modules/audio_processing/level_estimator_impl.cc
+++ b/modules/audio_processing/level_estimator_impl.cc
@@ -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 @@
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 @@
}
// 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 @@
}
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 @@
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);
diff --git a/modules/audio_processing/voice_detection_impl.cc b/modules/audio_processing/voice_detection_impl.cc
index 1d3d124..c6e497f 100644
--- a/modules/audio_processing/voice_detection_impl.cc
+++ b/modules/audio_processing/voice_detection_impl.cc
@@ -61,7 +61,7 @@
}
assert(audio->samples_per_split_channel() <= 160);
- int16_t* mixed_data = audio->low_pass_split_data(0);
+ const int16_t* mixed_data = audio->low_pass_split_data(0);
if (audio->num_channels() > 1) {
audio->CopyAndMixLowPass(1);
mixed_data = audio->mixed_low_pass_data(0);