mirror of
https://github.com/alsa-project/alsa-utils
synced 2024-11-08 23:15:41 +01:00
cba9050edf
Add common definitions of macros and data structures; Add functions that used by multiple components, such as wav file reading and writing. Signed-off-by: Lu, Han <han.lu@intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood@intel.com> Signed-off-by: Bernard Gautier <bernard.gautier@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
176 lines
4.7 KiB
C
176 lines
4.7 KiB
C
/*
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* Copyright (C) 2013-2015 Intel Corporation
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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*/
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#include <alsa/asoundlib.h>
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#define TEMP_RECORD_FILE_NAME "/tmp/bat.wav"
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#define OPT_BASE 300
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#define OPT_LOG (OPT_BASE + 1)
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#define OPT_READFILE (OPT_BASE + 2)
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#define OPT_SAVEPLAY (OPT_BASE + 3)
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#define OPT_LOCAL (OPT_BASE + 4)
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#define COMPOSE(a, b, c, d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24))
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#define WAV_RIFF COMPOSE('R', 'I', 'F', 'F')
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#define WAV_WAVE COMPOSE('W', 'A', 'V', 'E')
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#define WAV_FMT COMPOSE('f', 'm', 't', ' ')
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#define WAV_DATA COMPOSE('d', 'a', 't', 'a')
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#define WAV_FORMAT_PCM 1 /* PCM WAVE file encoding */
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#define MAX_CHANNELS 2
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#define MIN_CHANNELS 1
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#define MAX_PEAKS 10
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#define MAX_FRAMES (10 * 1024 * 1024)
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/* Given in ms */
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#define CAPTURE_DELAY 500
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/* signal frequency should be less than samplerate * RATE_FACTOR */
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#define RATE_FACTOR 0.4
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/* valid range of samplerate: (1 - RATE_RANGE, 1 + RATE_RANGE) * samplerate */
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#define RATE_RANGE 0.05
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/* Given in us */
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#define MAX_BUFFERTIME 500000
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/* devide factor, was 4, changed to 8 to remove reduce capture overrun */
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#define DIV_BUFFERTIME 8
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/* margin to avoid sign inversion when generate sine wav */
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#define RANGE_FACTOR 0.95
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#define EBATBASE 1000
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#define ENOPEAK (EBATBASE + 1)
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#define EONLYDC (EBATBASE + 2)
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#define EBADPEAK (EBATBASE + 3)
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#define DC_THRESHOLD 7.01
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/* tolerance of detected peak = max (DELTA_HZ, DELTA_RATE * target_freq).
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* If DELTA_RATE is too high, BAT may not be able to recognize negative result;
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* if too low, BAT may be too sensitive and results in uncecessary failure. */
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#define DELTA_RATE 0.005
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#define DELTA_HZ 1
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#define FOUND_DC (1<<1)
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#define FOUND_WRONG_PEAK (1<<0)
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struct wav_header {
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unsigned int magic; /* 'RIFF' */
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unsigned int length; /* file len */
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unsigned int type; /* 'WAVE' */
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};
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struct wav_chunk_header {
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unsigned int type; /* 'data' */
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unsigned int length; /* sample count */
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};
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struct wav_fmt {
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unsigned int magic; /* 'FMT '*/
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unsigned int fmt_size; /* 16 or 18 */
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unsigned short format; /* see WAV_FMT_* */
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unsigned short channels;
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unsigned int sample_rate; /* Frequency of sample */
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unsigned int bytes_p_second;
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unsigned short blocks_align; /* sample size; 1 or 2 bytes */
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unsigned short sample_length; /* 8, 12 or 16 bit */
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};
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struct chunk_fmt {
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unsigned short format; /* see WAV_FMT_* */
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unsigned short channels;
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unsigned int sample_rate; /* Frequency of sample */
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unsigned int bytes_p_second;
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unsigned short blocks_align; /* sample size; 1 or 2 bytes */
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unsigned short sample_length; /* 8, 12 or 16 bit */
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};
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struct wav_container {
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struct wav_header header;
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struct wav_fmt format;
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struct wav_chunk_header chunk;
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};
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struct bat;
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enum _bat_op_mode {
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MODE_UNKNOWN = -1,
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MODE_SINGLE = 0,
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MODE_LOOPBACK,
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MODE_LAST
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};
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struct pcm {
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char *device;
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char *file;
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enum _bat_op_mode mode;
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void *(*fct)(struct bat *);
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};
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struct sin_generator;
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struct sin_generator {
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double state_real;
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double state_imag;
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double phasor_real;
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double phasor_imag;
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float frequency;
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float sample_rate;
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float magnitude;
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};
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struct bat {
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unsigned int rate; /* sampling rate */
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int channels; /* nb of channels */
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int frames; /* nb of frames */
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int frame_size; /* size of frame */
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int sample_size; /* size of sample */
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snd_pcm_format_t format; /* PCM format */
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float sigma_k; /* threshold for peak detection */
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float target_freq[MAX_CHANNELS];
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int sinus_duration; /* number of frames for playback */
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char *narg; /* argument string of duration */
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char *logarg; /* path name of log file */
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char *debugplay; /* path name to store playback signal */
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struct pcm playback;
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struct pcm capture;
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unsigned int periods_played;
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unsigned int periods_total;
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bool period_is_limited;
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FILE *fp;
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FILE *log;
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FILE *err;
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void (*convert_sample_to_double)(void *, double *, int);
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void (*convert_float_to_sample)(float *, void *, int, int);
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void *buf; /* PCM Buffer */
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bool local; /* true for internal test */
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};
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struct analyze {
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void *buf;
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double *in;
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double *out;
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double *mag;
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};
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void prepare_wav_info(struct wav_container *, struct bat *);
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int read_wav_header(struct bat *, char *, FILE *, bool);
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int write_wav_header(FILE *, struct wav_container *, struct bat *);
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