WebRtc_Word32 -> int32_t in audio_processing/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index aee9f68..db1f2eb 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -54,10 +54,10 @@
int16_t low_pass_data[kSamplesPer16kHzChannel];
int16_t high_pass_data[kSamplesPer16kHzChannel];
- WebRtc_Word32 analysis_filter_state1[6];
- WebRtc_Word32 analysis_filter_state2[6];
- WebRtc_Word32 synthesis_filter_state1[6];
- WebRtc_Word32 synthesis_filter_state2[6];
+ int32_t analysis_filter_state1[6];
+ int32_t analysis_filter_state2[6];
+ int32_t synthesis_filter_state1[6];
+ int32_t synthesis_filter_state2[6];
};
// TODO(andrew): check range of input parameters?
@@ -142,22 +142,22 @@
return low_pass_reference_channels_[channel].data;
}
-WebRtc_Word32* AudioBuffer::analysis_filter_state1(int channel) const {
+int32_t* AudioBuffer::analysis_filter_state1(int channel) const {
assert(channel >= 0 && channel < num_channels_);
return split_channels_[channel].analysis_filter_state1;
}
-WebRtc_Word32* AudioBuffer::analysis_filter_state2(int channel) const {
+int32_t* AudioBuffer::analysis_filter_state2(int channel) const {
assert(channel >= 0 && channel < num_channels_);
return split_channels_[channel].analysis_filter_state2;
}
-WebRtc_Word32* AudioBuffer::synthesis_filter_state1(int channel) const {
+int32_t* AudioBuffer::synthesis_filter_state1(int channel) const {
assert(channel >= 0 && channel < num_channels_);
return split_channels_[channel].synthesis_filter_state1;
}
-WebRtc_Word32* AudioBuffer::synthesis_filter_state2(int channel) const {
+int32_t* AudioBuffer::synthesis_filter_state2(int channel) const {
assert(channel >= 0 && channel < num_channels_);
return split_channels_[channel].synthesis_filter_state2;
}