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Add two widely used constants
#define DEFAULT_SAMPLE_RATE 8000 #define DEFAULT_SAMPLES_PER_MS ((DEFAULT_SAMPLE_RATE)/1000) to the main header, and remove equivalent ones from plc.[ch] This will simplify the cleanup of the codec/ and formats/ files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -20,6 +20,9 @@
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#define DEFAULT_LANGUAGE "en"
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#define DEFAULT_LANGUAGE "en"
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#define DEFAULT_SAMPLE_RATE 8000
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#define DEFAULT_SAMPLES_PER_MS ((DEFAULT_SAMPLE_RATE)/1000)
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#define AST_CONFIG_MAX_PATH 255
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#define AST_CONFIG_MAX_PATH 255
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/* provided in asterisk.c */
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/* provided in asterisk.c */
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@@ -95,8 +95,6 @@ When a real packet is not available in time, call plc_fillin() to create a sythe
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That's it!
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That's it!
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*/
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*/
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#define SAMPLE_RATE 8000
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/*! Minimum allowed pitch (66 Hz) */
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/*! Minimum allowed pitch (66 Hz) */
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#define PLC_PITCH_MIN 120
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#define PLC_PITCH_MIN 120
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/*! Maximum allowed pitch (200 Hz) */
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/*! Maximum allowed pitch (200 Hz) */
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2
plc.c
2
plc.c
@@ -56,7 +56,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
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/* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */
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#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
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#define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */
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#define ms_to_samples(t) (((t)*SAMPLE_RATE)/1000)
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#define ms_to_samples(t) (((t)*DEFAULT_SAMPLE_RATE)/1000)
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static inline int16_t fsaturate(double damp)
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static inline int16_t fsaturate(double damp)
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{
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{
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