// // This file is part of Dire Wolf, an amateur radio packet TNC. // // Copyright (C) 2011, 2012, 2013, 2014, 2015 John Langner, WB2OSZ // // This program is free software: you can redistribute it and/or modify // it under the terms of the GNU General Public License as published by // the Free Software Foundation, either version 2 of the License, or // (at your option) any later version. // // This program is distributed in the hope that it will be useful, // but WITHOUT ANY WARRANTY; without even the implied warranty of // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the // GNU General Public License for more details. // // You should have received a copy of the GNU General Public License // along with this program. If not, see . // /*------------------------------------------------------------------ * * Module: audio.c * * Purpose: Interface to audio device commonly called a "sound card" for * historical reasons. * * This version is for Linux and Cygwin. * * Two different types of sound interfaces are supported: * * * OSS - For Cygwin or Linux versions with /dev/dsp. * * * ALSA - For Linux versions without /dev/dsp. * In this case, define preprocessor symbol USE_ALSA. * * References: Some tips on on using Linux sound devices. * * http://www.oreilly.de/catalog/multilinux/excerpt/ch14-05.htm * http://cygwin.com/ml/cygwin-patches/2004-q1/msg00116/devdsp.c * http://manuals.opensound.com/developer/fulldup.c.html * * "Introduction to Sound Programming with ALSA" * http://www.linuxjournal.com/article/6735?page=0,1 * * http://www.alsa-project.org/main/index.php/Asoundrc * * Credits: Release 1.0: Fabrice FAURE contributed code for the SDR UDP interface. * * Discussion here: http://gqrx.dk/doc/streaming-audio-over-udp * * Release 1.1: Gabor Berczi provided fixes for the OSS code * which had fallen into decay. * * Major Revisions: * * 1.2 - Add ability to use more than one audio device. * *---------------------------------------------------------------*/ #include "direwolf.h" #include #include #include #include #include #include #include #include #include #include #include #include #if USE_ALSA #include #else #include #ifdef __OpenBSD__ #include #else #include #endif #endif #include "audio.h" #include "audio_stats.h" #include "textcolor.h" #include "dtime_now.h" #include "demod.h" /* for alevel_t & demod_get_audio_level() */ /* Audio configuration. */ static struct audio_s *save_audio_config_p; /* Current state for each of the audio devices. */ static struct adev_s { #if USE_ALSA snd_pcm_t *audio_in_handle; snd_pcm_t *audio_out_handle; int bytes_per_frame; /* number of bytes for a sample from all channels. */ /* e.g. 4 for stereo 16 bit. */ #else int oss_audio_device_fd; /* Single device, both directions. */ #endif int inbuf_size_in_bytes; /* number of bytes allocated */ unsigned char *inbuf_ptr; int inbuf_len; /* number byte of actual data available. */ int inbuf_next; /* index of next to remove. */ int outbuf_size_in_bytes; unsigned char *outbuf_ptr; int outbuf_len; enum audio_in_type_e g_audio_in_type; int udp_sock; /* UDP socket for receiving data */ } adev[MAX_ADEVS]; // Originally 40. Version 1.2, try 10 for lower latency. #define ONE_BUF_TIME 10 #if USE_ALSA static int set_alsa_params (int a, snd_pcm_t *handle, struct audio_s *pa, char *name, char *dir); //static void alsa_select_device (char *pick_dev, int direction, char *result); #else static int set_oss_params (int a, int fd, struct audio_s *pa); #endif #define roundup1k(n) (((n) + 0x3ff) & ~0x3ff) static int calcbufsize(int rate, int chans, int bits) { int size1 = (rate * chans * bits / 8 * ONE_BUF_TIME) / 1000; int size2 = roundup1k(size1); #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_open: calcbufsize (rate=%d, chans=%d, bits=%d) calc size=%d, round up to %d\n", rate, chans, bits, size1, size2); #endif return (size2); } /*------------------------------------------------------------------ * * Name: audio_open * * Purpose: Open the digital audio device. * For "OSS", the device name is typically "/dev/dsp". * For "ALSA", it's a lot more complicated. See User Guide. * * New in version 1.0, we recognize "udp:" optionally * followed by a port number. * * Inputs: pa - Address of structure of type audio_s. * * Using a structure, rather than separate arguments * seemed to make sense because we often pass around * the same set of parameters various places. * * The fields that we care about are: * num_channels * samples_per_sec * bits_per_sample * If zero, reasonable defaults will be provided. * * The device names are in adevice_in and adevice_out. * - For "OSS", the device name is typically "/dev/dsp". * - For "ALSA", the device names are hw:c,d * where c is the "card" (for historical purposes) * and d is the "device" within the "card." * * * Outputs: pa - The ACTUAL values are returned here. * * These might not be exactly the same as what was requested. * * Example: ask for stereo, 16 bits, 22050 per second. * An ordinary desktop/laptop PC should be able to handle this. * However, some other sort of smaller device might be * more restrictive in its capabilities. * It might say, the best I can do is mono, 8 bit, 8000/sec. * * The sofware modem must use this ACTUAL information * that the device is supplying, that could be different * than what the user specified. * * Returns: 0 for success, -1 for failure. * * *----------------------------------------------------------------*/ int audio_open (struct audio_s *pa) { int err; int chan; int a; char audio_in_name[30]; char audio_out_name[30]; save_audio_config_p = pa; memset (adev, 0, sizeof(adev)); for (a=0; aadev[a].num_channels == 0) pa->adev[a].num_channels = DEFAULT_NUM_CHANNELS; if (pa->adev[a].samples_per_sec == 0) pa->adev[a].samples_per_sec = DEFAULT_SAMPLES_PER_SEC; if (pa->adev[a].bits_per_sample == 0) pa->adev[a].bits_per_sample = DEFAULT_BITS_PER_SAMPLE; for (chan=0; chanachan[chan].mark_freq == 0) pa->achan[chan].mark_freq = DEFAULT_MARK_FREQ; if (pa->achan[chan].space_freq == 0) pa->achan[chan].space_freq = DEFAULT_SPACE_FREQ; if (pa->achan[chan].baud == 0) pa->achan[chan].baud = DEFAULT_BAUD; if (pa->achan[chan].num_subchan == 0) pa->achan[chan].num_subchan = 1; } } /* * Open audio device(s). */ for (a=0; aadev[a].defined) { adev[a].inbuf_size_in_bytes = 0; adev[a].inbuf_ptr = NULL; adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; adev[a].outbuf_size_in_bytes = 0; adev[a].outbuf_ptr = NULL; adev[a].outbuf_len = 0; /* * Determine the type of audio input. */ adev[a].g_audio_in_type = AUDIO_IN_TYPE_SOUNDCARD; if (strcasecmp(pa->adev[a].adevice_in, "stdin") == 0 || strcmp(pa->adev[a].adevice_in, "-") == 0) { adev[a].g_audio_in_type = AUDIO_IN_TYPE_STDIN; /* Change "-" to stdin for readability. */ strlcpy (pa->adev[a].adevice_in, "stdin", sizeof(pa->adev[a].adevice_in)); } if (strncasecmp(pa->adev[a].adevice_in, "udp:", 4) == 0) { adev[a].g_audio_in_type = AUDIO_IN_TYPE_SDR_UDP; /* Supply default port if none specified. */ if (strcasecmp(pa->adev[a].adevice_in,"udp") == 0 || strcasecmp(pa->adev[a].adevice_in,"udp:") == 0) { snprintf (pa->adev[a].adevice_in, sizeof(pa->adev[a].adevice_in), "udp:%d", DEFAULT_UDP_AUDIO_PORT); } } /* Let user know what is going on. */ /* If not specified, the device names should be "default". */ strlcpy (audio_in_name, pa->adev[a].adevice_in, sizeof(audio_in_name)); strlcpy (audio_out_name, pa->adev[a].adevice_out, sizeof(audio_out_name)); char ctemp[40]; if (pa->adev[a].num_channels == 2) { snprintf (ctemp, sizeof(ctemp), " (channels %d & %d)", ADEVFIRSTCHAN(a), ADEVFIRSTCHAN(a)+1); } else { snprintf (ctemp, sizeof(ctemp), " (channel %d)", ADEVFIRSTCHAN(a)); } text_color_set(DW_COLOR_INFO); if (strcmp(audio_in_name,audio_out_name) == 0) { dw_printf ("Audio device for both receive and transmit: %s %s\n", audio_in_name, ctemp); } else { dw_printf ("Audio input device for receive: %s %s\n", audio_in_name, ctemp); dw_printf ("Audio out device for transmit: %s %s\n", audio_out_name, ctemp); } /* * Now attempt actual opens. */ /* * Input device. */ switch (adev[a].g_audio_in_type) { /* * Soundcard - ALSA. */ case AUDIO_IN_TYPE_SOUNDCARD: #if USE_ALSA err = snd_pcm_open (&(adev[a].audio_in_handle), audio_in_name, SND_PCM_STREAM_CAPTURE, 0); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not open audio device %s for input\n%s\n", audio_in_name, snd_strerror(err)); return (-1); } adev[a].inbuf_size_in_bytes = set_alsa_params (a, adev[a].audio_in_handle, pa, audio_in_name, "input"); #else // OSS adev[a].oss_audio_device_fd = open (pa->adev[a].adevice_in, O_RDWR); if (adev[a].oss_audio_device_fd < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("%s:\n", pa->adev[a].adevice_in); // snprintf (message, sizeof(message), "Could not open audio device %s", pa->adev[a].adevice_in); // perror (message); return (-1); } adev[a].outbuf_size_in_bytes = adev[a].inbuf_size_in_bytes = set_oss_params (a, adev[a].oss_audio_device_fd, pa); if (adev[a].inbuf_size_in_bytes <= 0 || adev[a].outbuf_size_in_bytes <= 0) { return (-1); } #endif break; /* * UDP. */ case AUDIO_IN_TYPE_SDR_UDP: //Create socket and bind socket { struct sockaddr_in si_me; //int slen=sizeof(si_me); //int data_size = 0; //Create UDP Socket if ((adev[a].udp_sock=socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP))==-1) { text_color_set(DW_COLOR_ERROR); dw_printf ("Couldn't create socket, errno %d\n", errno); return -1; } memset((char *) &si_me, 0, sizeof(si_me)); si_me.sin_family = AF_INET; si_me.sin_port = htons((short)atoi(audio_in_name+4)); si_me.sin_addr.s_addr = htonl(INADDR_ANY); //Bind to the socket if (bind(adev[a].udp_sock, (const struct sockaddr *) &si_me, sizeof(si_me))==-1) { text_color_set(DW_COLOR_ERROR); dw_printf ("Couldn't bind socket, errno %d\n", errno); return -1; } } adev[a].inbuf_size_in_bytes = SDR_UDP_BUF_MAXLEN; break; /* * stdin. */ case AUDIO_IN_TYPE_STDIN: /* Do we need to adjust any properties of stdin? */ adev[a].inbuf_size_in_bytes = 1024; break; default: text_color_set(DW_COLOR_ERROR); dw_printf ("Internal error, invalid audio_in_type\n"); return (-1); } /* * Output device. Only "soundcard" is supported at this time. */ #if USE_ALSA err = snd_pcm_open (&(adev[a].audio_out_handle), audio_out_name, SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not open audio device %s for output\n%s\n", audio_out_name, snd_strerror(err)); return (-1); } adev[a].outbuf_size_in_bytes = set_alsa_params (a, adev[a].audio_out_handle, pa, audio_out_name, "output"); if (adev[a].inbuf_size_in_bytes <= 0 || adev[a].outbuf_size_in_bytes <= 0) { return (-1); } #endif /* * Finally allocate buffer for each direction. */ adev[a].inbuf_ptr = malloc(adev[a].inbuf_size_in_bytes); assert (adev[a].inbuf_ptr != NULL); adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; adev[a].outbuf_ptr = malloc(adev[a].outbuf_size_in_bytes); assert (adev[a].outbuf_ptr != NULL); adev[a].outbuf_len = 0; } /* end of audio device defined */ } /* end of for each audio device */ return (0); } /* end audio_open */ #if USE_ALSA /* * Set parameters for sound card. * * See ?? for details. */ /* * Terminology: * Sample - for one channel. e.g. 2 bytes for 16 bit. * Frame - one sample for all channels. e.g. 4 bytes for 16 bit stereo * Period - size of one transfer. */ static int set_alsa_params (int a, snd_pcm_t *handle, struct audio_s *pa, char *devname, char *inout) { snd_pcm_hw_params_t *hw_params; snd_pcm_uframes_t fpp; /* Frames per period. */ unsigned int val; int dir; int err; int buf_size_in_bytes; /* result, number of bytes per transfer. */ err = snd_pcm_hw_params_malloc (&hw_params); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not alloc hw param structure.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } err = snd_pcm_hw_params_any (handle, hw_params); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not init hw param structure.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } /* Interleaved data: L, R, L, R, ... */ err = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set interleaved mode.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } /* Signed 16 bit little endian or unsigned 8 bit. */ err = snd_pcm_hw_params_set_format (handle, hw_params, pa->adev[a].bits_per_sample == 8 ? SND_PCM_FORMAT_U8 : SND_PCM_FORMAT_S16_LE); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set bits per sample.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } /* Number of audio channels. */ err = snd_pcm_hw_params_set_channels (handle, hw_params, pa->adev[a].num_channels); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set number of audio channels.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } /* Audio sample rate. */ val = pa->adev[a].samples_per_sec; dir = 0; err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &val, &dir); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set audio sample rate.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } if (val != pa->adev[a].samples_per_sec) { text_color_set(DW_COLOR_INFO); dw_printf ("Asked for %d samples/sec but got %d.\n", pa->adev[a].samples_per_sec, val); dw_printf ("for %s %s.\n", devname, inout); pa->adev[a].samples_per_sec = val; } /* Original: */ /* Guessed around 20 reads/sec might be good. */ /* Period too long = too much latency. */ /* Period too short = more overhead of many small transfers. */ /* fpp = pa->adev[a].samples_per_sec / 20; */ /* The suggested period size was 2205 frames. */ /* I thought the later "...set_period_size_near" might adjust it to be */ /* some more optimal nearby value based hardware buffer sizes but */ /* that didn't happen. We ended up with a buffer size of 4410 bytes. */ /* In version 1.2, let's take a different approach. */ /* Reduce the latency and round up to a multiple of 1 Kbyte. */ /* For the typical case of 44100 sample rate, 1 channel, 16 bits, we calculate */ /* a buffer size of 882 and round it up to 1k. This results in 512 frames per period. */ /* A period comes out to be about 80 periods per second or about 12.5 mSec each. */ buf_size_in_bytes = calcbufsize(pa->adev[a].samples_per_sec, pa->adev[a].num_channels, pa->adev[a].bits_per_sample); #if __arm__ /* Ugly hack for RPi. */ /* Reducing buffer size is fine for input but not so good for output. */ if (*inout == 'o') { buf_size_in_bytes = buf_size_in_bytes * 4; } #endif fpp = buf_size_in_bytes / (pa->adev[a].num_channels * pa->adev[a].bits_per_sample / 8); #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("suggest period size of %d frames\n", (int)fpp); #endif dir = 0; err = snd_pcm_hw_params_set_period_size_near (handle, hw_params, &fpp, &dir); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set period size\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } err = snd_pcm_hw_params (handle, hw_params); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not set hw params\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } /* Driver might not like our suggested period size */ /* and might have another idea. */ err = snd_pcm_hw_params_get_period_size (hw_params, &fpp, NULL); if (err < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Could not get audio period size.\n%s\n", snd_strerror(err)); dw_printf ("for %s %s.\n", devname, inout); return (-1); } snd_pcm_hw_params_free (hw_params); /* A "frame" is one sample for all channels. */ /* The read and write use units of frames, not bytes. */ adev[a].bytes_per_frame = snd_pcm_frames_to_bytes (handle, 1); assert (adev[a].bytes_per_frame == pa->adev[a].num_channels * pa->adev[a].bits_per_sample / 8); buf_size_in_bytes = fpp * adev[a].bytes_per_frame; #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio buffer size = %d (bytes per frame) x %d (frames per period) = %d \n", adev[a].bytes_per_frame, (int)fpp, buf_size_in_bytes); #endif /* Version 1.3 - after a report of this situation for Mac OSX version. */ if (buf_size_in_bytes < 256 || buf_size_in_bytes > 32768) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio buffer has unexpected extreme size of %d bytes.\n", buf_size_in_bytes); dw_printf ("Detected at %s, line %d.\n", __FILE__, __LINE__); dw_printf ("This might be caused by unusual audio device configuration values.\n"); buf_size_in_bytes = 2048; dw_printf ("Using %d to attempt recovery.\n", buf_size_in_bytes); } return (buf_size_in_bytes); } /* end alsa_set_params */ #else /* * Set parameters for sound card. (OSS only) * * See /usr/include/sys/soundcard.h for details. */ static int set_oss_params (int a, int fd, struct audio_s *pa) { int err; int devcaps; int asked_for; char message[100]; int ossbuf_size_in_bytes; err = ioctl (fd, SNDCTL_DSP_CHANNELS, &(pa->adev[a].num_channels)); if (err == -1) { text_color_set(DW_COLOR_ERROR); perror("Not able to set audio device number of channels"); return (-1); } asked_for = pa->adev[a].samples_per_sec; err = ioctl (fd, SNDCTL_DSP_SPEED, &(pa->adev[a].samples_per_sec)); if (err == -1) { text_color_set(DW_COLOR_ERROR); perror("Not able to set audio device sample rate"); return (-1); } if (pa->adev[a].samples_per_sec != asked_for) { text_color_set(DW_COLOR_INFO); dw_printf ("Asked for %d samples/sec but actually using %d.\n", asked_for, pa->adev[a].samples_per_sec); } /* This is actually a bit mask but it happens that */ /* 0x8 is unsigned 8 bit samples and */ /* 0x10 is signed 16 bit little endian. */ err = ioctl (fd, SNDCTL_DSP_SETFMT, &(pa->adev[a].bits_per_sample)); if (err == -1) { text_color_set(DW_COLOR_ERROR); perror("Not able to set audio device sample size"); return (-1); } /* * Determine capabilities. */ err = ioctl (fd, SNDCTL_DSP_GETCAPS, &devcaps); if (err == -1) { text_color_set(DW_COLOR_ERROR); perror("Not able to get audio device capabilities"); // Is this fatal? // return (-1); } #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_open(): devcaps = %08x\n", devcaps); if (devcaps & DSP_CAP_DUPLEX) dw_printf ("Full duplex record/playback.\n"); if (devcaps & DSP_CAP_BATCH) dw_printf ("Device has some kind of internal buffers which may cause delays.\n"); if (devcaps & ~ (DSP_CAP_DUPLEX | DSP_CAP_BATCH)) dw_printf ("Others...\n"); #endif if (!(devcaps & DSP_CAP_DUPLEX)) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio device does not support full duplex\n"); // Do we care? // return (-1); } err = ioctl (fd, SNDCTL_DSP_SETDUPLEX, NULL); if (err == -1) { // text_color_set(DW_COLOR_ERROR); // perror("Not able to set audio full duplex mode"); // Unfortunate but not a disaster. } /* * Get preferred block size. * Presumably this will provide the most efficient transfer. * * In my particular situation, this turned out to be * 2816 for 11025 Hz 16 bit mono * 5568 for 11025 Hz 16 bit stereo * 11072 for 44100 Hz 16 bit mono * * This was long ago under different conditions. * Should study this again some day. * * Your milage may vary. */ err = ioctl (fd, SNDCTL_DSP_GETBLKSIZE, &ossbuf_size_in_bytes); if (err == -1) { text_color_set(DW_COLOR_ERROR); perror("Not able to get audio block size"); ossbuf_size_in_bytes = 2048; /* pick something reasonable */ } #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_open(): suggestd block size is %d\n", ossbuf_size_in_bytes); #endif /* * That's 1/8 of a second which seems rather long if we want to * respond quickly. */ ossbuf_size_in_bytes = calcbufsize(pa->adev[a].samples_per_sec, pa->adev[a].num_channels, pa->adev[a].bits_per_sample); #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_open(): using block size of %d\n", ossbuf_size_in_bytes); #endif #if 0 /* Original - dies without good explanation. */ assert (ossbuf_size_in_bytes >= 256 && ossbuf_size_in_bytes <= 32768); #else /* Version 1.3 - after a report of this situation for Mac OSX version. */ if (ossbuf_size_in_bytes < 256 || ossbuf_size_in_bytes > 32768) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio buffer has unexpected extreme size of %d bytes.\n", ossbuf_size_in_bytes); dw_printf ("Detected at %s, line %d.\n", __FILE__, __LINE__); dw_printf ("This might be caused by unusual audio device configuration values.\n"); ossbuf_size_in_bytes = 2048; dw_printf ("Using %d to attempt recovery.\n", ossbuf_size_in_bytes); } #endif return (ossbuf_size_in_bytes); } /* end set_oss_params */ #endif /*------------------------------------------------------------------ * * Name: audio_get * * Purpose: Get one byte from the audio device. * * Inputs: a - Our number for audio device. * * Returns: 0 - 255 for a valid sample. * -1 for any type of error. * * Description: The caller must deal with the details of mono/stereo * and number of bytes per sample. * * This will wait if no data is currently available. * *----------------------------------------------------------------*/ // Use hot attribute for all functions called for every audio sample. __attribute__((hot)) int audio_get (int a) { int n; int retries = 0; #if STATISTICS /* Gather numbers for read from audio device. */ #define duration 100 /* report every 100 seconds. */ static time_t last_time[MAX_ADEVS]; time_t this_time[MAX_ADEVS]; static int sample_count[MAX_ADEVS]; static int error_count[MAX_ADEVS]; #endif #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_get():\n"); #endif assert (adev[a].inbuf_size_in_bytes >= 100 && adev[a].inbuf_size_in_bytes <= 32768); switch (adev[a].g_audio_in_type) { /* * Soundcard - ALSA */ case AUDIO_IN_TYPE_SOUNDCARD: #if USE_ALSA while (adev[a].inbuf_next >= adev[a].inbuf_len) { assert (adev[a].audio_in_handle != NULL); #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_get(): readi asking for %d frames\n", adev[a].inbuf_size_in_bytes / adev[a].bytes_per_frame); #endif n = snd_pcm_readi (adev[a].audio_in_handle, adev[a].inbuf_ptr, adev[a].inbuf_size_in_bytes / adev[a].bytes_per_frame); #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_get(): readi asked for %d and got %d frames\n", adev[a].inbuf_size_in_bytes / adev[a].bytes_per_frame, n); #endif if (n > 0) { /* Success */ adev[a].inbuf_len = n * adev[a].bytes_per_frame; /* convert to number of bytes */ adev[a].inbuf_next = 0; audio_stats (a, save_audio_config_p->adev[a].num_channels, n, save_audio_config_p->statistics_interval); } else if (n == 0) { /* Didn't expect this, but it's not a problem. */ /* Wait a little while and try again. */ text_color_set(DW_COLOR_ERROR); dw_printf ("Audio input got zero bytes: %s\n", snd_strerror(n)); SLEEP_MS(10); adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; } else { /* Error */ // TODO: Needs more study and testing. // Only expected error conditions: // -EBADFD PCM is not in the right state (SND_PCM_STATE_PREPARED or SND_PCM_STATE_RUNNING) // -EPIPE an overrun occurred // -ESTRPIPE a suspend event occurred (stream is suspended and waiting for an application recovery) // Data overrun is displayed as "broken pipe" which seems a little misleading. // Add our own message which says something about CPU being too slow. text_color_set(DW_COLOR_ERROR); dw_printf ("Audio input device %d error code %d: %s\n", a, n, snd_strerror(n)); if (n == (-EPIPE)) { dw_printf ("This is most likely caused by the CPU being too slow to keep up with the audio stream.\n"); dw_printf ("Use the \"top\" command, in another command window, to look at CPU usage.\n"); dw_printf ("This might be a temporary condition so we will attempt to recover a few times before giving up.\n"); } audio_stats (a, save_audio_config_p->adev[a].num_channels, 0, save_audio_config_p->statistics_interval); /* Try to recover a few times and eventually give up. */ if (++retries > 10) { adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; return (-1); } if (n == -EPIPE) { /* EPIPE means overrun */ snd_pcm_recover (adev[a].audio_in_handle, n, 1); } else { /* Could be some temporary condition. */ /* Wait a little then try again. */ /* Sometimes I get "Resource temporarily available" */ /* when the Update Manager decides to run. */ SLEEP_MS (250); snd_pcm_recover (adev[a].audio_in_handle, n, 1); } } } #else /* end ALSA, begin OSS */ /* Fixed in 1.2. This was formerly outside of the switch */ /* so the OSS version did not process stdin or UDP. */ while (adev[a].g_audio_in_type == AUDIO_IN_TYPE_SOUNDCARD && adev[a].inbuf_next >= adev[a].inbuf_len) { assert (adev[a].oss_audio_device_fd > 0); n = read (adev[a].oss_audio_device_fd, adev[a].inbuf_ptr, adev[a].inbuf_size_in_bytes); //text_color_set(DW_COLOR_DEBUG); // dw_printf ("audio_get(): read %d returns %d\n", adev[a].inbuf_size_in_bytes, n); if (n < 0) { text_color_set(DW_COLOR_ERROR); perror("Can't read from audio device"); adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; audio_stats (a, save_audio_config_p->adev[a].num_channels, 0, save_audio_config_p->statistics_interval); return (-1); } adev[a].inbuf_len = n; adev[a].inbuf_next = 0; audio_stats (a, save_audio_config_p->adev[a].num_channels, n / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8), save_audio_config_p->statistics_interval); } #endif /* USE_ALSA */ break; /* * UDP. */ case AUDIO_IN_TYPE_SDR_UDP: while (adev[a].inbuf_next >= adev[a].inbuf_len) { int res; assert (adev[a].udp_sock > 0); res = recv(adev[a].udp_sock, adev[a].inbuf_ptr, adev[a].inbuf_size_in_bytes, 0); if (res < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Can't read from udp socket, res=%d", res); adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; audio_stats (a, save_audio_config_p->adev[a].num_channels, 0, save_audio_config_p->statistics_interval); return (-1); } adev[a].inbuf_len = res; adev[a].inbuf_next = 0; audio_stats (a, save_audio_config_p->adev[a].num_channels, res / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8), save_audio_config_p->statistics_interval); } break; /* * stdin. */ case AUDIO_IN_TYPE_STDIN: while (adev[a].inbuf_next >= adev[a].inbuf_len) { //int ch, res,i; int res; res = read(STDIN_FILENO, adev[a].inbuf_ptr, (size_t)adev[a].inbuf_size_in_bytes); if (res <= 0) { text_color_set(DW_COLOR_INFO); dw_printf ("\nEnd of file on stdin. Exiting.\n"); exit (0); } audio_stats (a, save_audio_config_p->adev[a].num_channels, res / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8), save_audio_config_p->statistics_interval); adev[a].inbuf_len = res; adev[a].inbuf_next = 0; } break; } if (adev[a].inbuf_next < adev[a].inbuf_len) n = adev[a].inbuf_ptr[adev[a].inbuf_next++]; //No data to read, avoid reading outside buffer else n = 0; #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_get(): returns %d\n", n); #endif return (n); } /* end audio_get */ /*------------------------------------------------------------------ * * Name: audio_put * * Purpose: Send one byte to the audio device. * * Inputs: a * * c - One byte in range of 0 - 255. * * Returns: Normally non-negative. * -1 for any type of error. * * Description: The caller must deal with the details of mono/stereo * and number of bytes per sample. * * See Also: audio_flush * audio_wait * *----------------------------------------------------------------*/ int audio_put (int a, int c) { /* Should never be full at this point. */ assert (adev[a].outbuf_len < adev[a].outbuf_size_in_bytes); adev[a].outbuf_ptr[adev[a].outbuf_len++] = c; if (adev[a].outbuf_len == adev[a].outbuf_size_in_bytes) { return (audio_flush(a)); } return (0); } /* end audio_put */ /*------------------------------------------------------------------ * * Name: audio_flush * * Purpose: Push out any partially filled output buffer. * * Returns: Normally non-negative. * -1 for any type of error. * * See Also: audio_flush * audio_wait * *----------------------------------------------------------------*/ int audio_flush (int a) { #if USE_ALSA int k; unsigned char *psound; int retries = 10; snd_pcm_status_t *status; assert (adev[a].audio_out_handle != NULL); /* * Trying to set the automatic start threshold didn't have the desired * effect. After the first transmitted packet, they are saved up * for a few minutes and then all come out together. * * "Prepare" it if not already in the running state. * We stop it at the end of each transmitted packet. */ snd_pcm_status_alloca(&status); k = snd_pcm_status (adev[a].audio_out_handle, status); if (k != 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio output get status error.\n%s\n", snd_strerror(k)); } if ((k = snd_pcm_status_get_state(status)) != SND_PCM_STATE_RUNNING) { //text_color_set(DW_COLOR_DEBUG); //dw_printf ("Audio output state = %d. Try to start.\n", k); k = snd_pcm_prepare (adev[a].audio_out_handle); if (k != 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio output start error.\n%s\n", snd_strerror(k)); } } psound = adev[a].outbuf_ptr; while (retries-- > 0) { k = snd_pcm_writei (adev[a].audio_out_handle, psound, adev[a].outbuf_len / adev[a].bytes_per_frame); #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_flush(): snd_pcm_writei %d frames returns %d\n", adev[a].outbuf_len / adev[a].bytes_per_frame, k); fflush (stdout); #endif if (k == -EPIPE) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio output data underrun.\n"); /* No problemo. Recover and go around again. */ snd_pcm_recover (adev[a].audio_out_handle, k, 1); } else if (k < 0) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio write error: %s\n", snd_strerror(k)); /* Some other error condition. */ /* Try again. What do we have to lose? */ snd_pcm_recover (adev[a].audio_out_handle, k, 1); } else if (k != adev[a].outbuf_len / adev[a].bytes_per_frame) { text_color_set(DW_COLOR_ERROR); dw_printf ("Audio write took %d frames rather than %d.\n", k, adev[a].outbuf_len / adev[a].bytes_per_frame); /* Go around again with the rest of it. */ psound += k * adev[a].bytes_per_frame; adev[a].outbuf_len -= k * adev[a].bytes_per_frame; } else { /* Success! */ adev[a].outbuf_len = 0; return (0); } } text_color_set(DW_COLOR_ERROR); dw_printf ("Audio write error retry count exceeded.\n"); adev[a].outbuf_len = 0; return (-1); #else /* OSS */ int k; unsigned char *ptr; int len; ptr = adev[a].outbuf_ptr; len = adev[a].outbuf_len; while (len > 0) { assert (adev[a].oss_audio_device_fd > 0); k = write (adev[a].oss_audio_device_fd, ptr, len); #if DEBUGx text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_flush(): write %d returns %d\n", len, k); fflush (stdout); #endif if (k < 0) { text_color_set(DW_COLOR_ERROR); perror("Can't write to audio device"); adev[a].outbuf_len = 0; return (-1); } if (k < len) { /* presumably full but didn't block. */ usleep (10000); } ptr += k; len -= k; } adev[a].outbuf_len = 0; return (0); #endif } /* end audio_flush */ /*------------------------------------------------------------------ * * Name: audio_wait * * Purpose: Finish up audio output before turning PTT off. * * Inputs: a - Index for audio device (not channel!) * * Returns: None. * * Description: Flush out any partially filled audio output buffer. * Wait until all the queued up audio out has been played. * Take any other necessary actions to stop audio output. * * In an ideal world: * * We would like to ask the hardware when all the queued * up sound has actually come out the speaker. * * In reality: * * This has been found to be less than reliable in practice. * * Caller does the following: * * (1) Make note of when PTT is turned on. * (2) Calculate how long it will take to transmit the * frame including TXDELAY, frame (including * "flags", data, FCS and bit stuffing), and TXTAIL. * (3) Call this function, which might or might not wait long enough. * (4) Add (1) and (2) resulting in when PTT should be turned off. * (5) Take difference between current time and desired PPT off time * and wait for additoinal time if required. * *----------------------------------------------------------------*/ void audio_wait (int a) { audio_flush (a); #if USE_ALSA /* For playback, this should wait for all pending frames */ /* to be played and then stop. */ snd_pcm_drain (adev[a].audio_out_handle); /* * When this was first implemented, I observed: * * "Experimentation reveals that snd_pcm_drain doesn't * actually wait. It returns immediately. * However it does serve a useful purpose of stopping * the playback after all the queued up data is used." * * * Now that I take a closer look at the transmit timing, for * version 1.2, it seems that snd_pcm_drain DOES wait until all * all pending frames have been played. * Either way, the caller will now compensate for it. */ #else assert (adev[a].oss_audio_device_fd > 0); // This caused a crash later on Cygwin. // Haven't tried it on other (non-Linux) Unix yet. // err = ioctl (adev[a].oss_audio_device_fd, SNDCTL_DSP_SYNC, NULL); #endif #if DEBUG text_color_set(DW_COLOR_DEBUG); dw_printf ("audio_wait(): after sync, status=%d\n", err); #endif } /* end audio_wait */ /*------------------------------------------------------------------ * * Name: audio_close * * Purpose: Close the audio device(s). * * Returns: Normally non-negative. * -1 for any type of error. * * *----------------------------------------------------------------*/ int audio_close (void) { int err = 0; int a; for (a = 0; a < MAX_ADEVS; a++) { #if USE_ALSA if (adev[a].audio_in_handle != NULL && adev[a].audio_out_handle != NULL) { audio_wait (a); snd_pcm_close (adev[a].audio_in_handle); snd_pcm_close (adev[a].audio_out_handle); #else if (adev[a].oss_audio_device_fd > 0) { audio_wait (a); close (adev[a].oss_audio_device_fd); adev[a].oss_audio_device_fd = -1; #endif free (adev[a].inbuf_ptr); free (adev[a].outbuf_ptr); adev[a].inbuf_size_in_bytes = 0; adev[a].inbuf_ptr = NULL; adev[a].inbuf_len = 0; adev[a].inbuf_next = 0; adev[a].outbuf_size_in_bytes = 0; adev[a].outbuf_ptr = NULL; adev[a].outbuf_len = 0; } } return (err); } /* end audio_close */ /* end audio.c */