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WebRtc_Word32 -> int32_t in audio_processing/

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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
This commit is contained in:
pbos@webrtc.org
2013-04-10 07:50:54 +00:00
parent 557e92515d
commit b7192b8247
37 changed files with 1189 additions and 1191 deletions
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16
webrtc/modules/audio_processing/audio_buffer.cc
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@ -54,10 +54,10 @@ struct SplitAudioChannel {
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 @@ int16_t* AudioBuffer::low_pass_reference(int channel) const {
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;
}