commit 65f933899b815b6c09f8ca7beefeace09ee3ae70
author andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Apr 30 16:44:13 2014 +0000
committer andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Apr 30 16:44:13 2014 +0000
tree 30a0075b6b1b6fa7dbceb71e31263d611c00aa0a
parent 740e6b339a070bd571a559f6d7aee4c604fd4c5e

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@6030 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_processing/audio_buffer.cc

diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index c53d4df..9160f69 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/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];
 }