From 1674096e45730cd41779cb8293faab4ec3422665 Mon Sep 17 00:00:00 2001 Message-Id: <1674096e45730cd41779cb8293faab4ec3422665.1714864445.git.mdw@distorted.org.uk> From: Mark Wooding Date: Sat, 22 Sep 2007 19:54:27 +0100 Subject: [PATCH 1/1] move audio translation to the right place Organization: Straylight/Edgeware From: rjk@greenend.org.uk <> --- server/speaker.c | 277 +++++++++++++++++++++++++++++------------------ 1 file changed, 170 insertions(+), 107 deletions(-) diff --git a/server/speaker.c b/server/speaker.c index acbc9bf..72c1618 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -17,14 +17,35 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 * USA */ - -/* This program deliberately does not use the garbage collector even though it - * might be convenient to do so. This is for two reasons. Firstly some libao - * drivers are implemented using threads and we do not want to have to deal - * with potential interactions between threading and garbage collection. - * Secondly this process needs to be able to respond quickly and this is not - * compatible with the collector hanging the program even relatively - * briefly. */ +/** @file server/speaker.c + * @brief Speaker processs + * + * This program is responsible for transmitting a single coherent audio stream + * to its destination (over the network, to some sound API, to some + * subprocess). It receives connections from decoders via file descriptor + * passing from the main server and plays them in the right order. + * + * For the ALSA API, 8- and 16- bit + * stereo and mono are supported, with any sample rate (within the limits that + * ALSA can deal with.) + * + * When communicating with a subprocess, sox is invoked to convert the inbound + * data to a single consistent format. The same applies for network (RTP) + * play, though in that case currently only 44.1KHz 16-bit stereo is supported. + * + * The inbound data starts with a structure defining the data format. Note + * that this is NOT portable between different platforms or even necessarily + * between versions; the speaker is assumed to be built from the same source + * and run on the same host as the main server. + * + * This program deliberately does not use the garbage collector even though it + * might be convenient to do so. This is for two reasons. Firstly some sound + * APIs use thread threads and we do not want to have to deal with potential + * interactions between threading and garbage collection. Secondly this + * process needs to be able to respond quickly and this is not compatible with + * the collector hanging the program even relatively briefly. + */ #include #include "types.h" @@ -70,20 +91,32 @@ # define MACHINE_AO_FMT AO_FMT_LITTLE #endif -#define BUFFER_SECONDS 5 /* How many seconds of input to - * buffer. */ +/** @brief How many seconds of input to buffer + * + * While any given connection has this much audio buffered, no more reads will + * be issued for that connection. The decoder will have to wait. + */ +#define BUFFER_SECONDS 5 #define FRAMES 4096 /* Frame batch size */ -#define NETWORK_BYTES 1024 /* Bytes to send per network packet */ -/* (don't make this too big or arithmetic will start to overflow) */ +/** @brief Bytes to send per network packet + * + * Don't make this too big or arithmetic will start to overflow. + */ +#define NETWORK_BYTES 1024 -#define RTP_AHEAD 2 /* Max RTP playahead (seconds) */ +/** @brief Maximum RTP playahead (seconds) */ +#define RTP_AHEAD 2 -#define NFDS 256 /* Max FDs to poll for */ +/** @brief Maximum number of FDs to poll for */ +#define NFDS 256 -/* Known tracks are kept in a linked list. We don't normally to have - * more than two - maybe three at the outside. */ +/** @brief Track structure + * + * Known tracks are kept in a linked list. Usually there will be at most two + * of these but rearranging the queue can cause there to be more. + */ static struct track { struct track *next; /* next track */ int fd; /* input FD */ @@ -152,12 +185,12 @@ static void version(void) { exit(0); } -/* Return the number of bytes per frame in FORMAT. */ +/** @brief Return the number of bytes per frame in @p format */ static size_t bytes_per_frame(const ao_sample_format *format) { return format->channels * format->bits / 8; } -/* Find track ID, maybe creating it if not found. */ +/** @brief Find track @p id, maybe creating it if not found */ static struct track *findtrack(const char *id, int create) { struct track *t; @@ -177,7 +210,7 @@ static struct track *findtrack(const char *id, int create) { return t; } -/* Remove track ID (but do not destroy it). */ +/** @brief Remove track @p id (but do not destroy it) */ static struct track *removetrack(const char *id) { struct track *t, **tt; @@ -189,7 +222,7 @@ static struct track *removetrack(const char *id) { return t; } -/* Destroy a track. */ +/** @brief Destroy a track */ static void destroy(struct track *t) { D(("destroy %s", t->id)); if(t->fd != -1) xclose(t->fd); @@ -197,7 +230,7 @@ static void destroy(struct track *t) { free(t); } -/* Notice a new FD. */ +/** @brief Notice a new connection */ static void acquire(struct track *t, int fd) { D(("acquire %s %d", t->id, fd)); if(t->fd != -1) @@ -206,7 +239,111 @@ static void acquire(struct track *t, int fd) { nonblock(fd); } -/* Read data into a sample buffer. Return 0 on success, -1 on EOF. */ +/** @brief Return true if A and B denote identical libao formats, else false */ +static int formats_equal(const ao_sample_format *a, + const ao_sample_format *b) { + return (a->bits == b->bits + && a->rate == b->rate + && a->channels == b->channels + && a->byte_format == b->byte_format); +} + +/** @brief Compute arguments to sox */ +static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { + int n; + + *(*pp)++ = "-t.raw"; + *(*pp)++ = "-s"; + *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; + *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; + /* sox 12.17.9 insists on -b etc; CVS sox insists on - etc; both are + * deployed! */ + switch(config->sox_generation) { + case 0: + if(ao->bits != 8 + && ao->byte_format != AO_FMT_NATIVE + && ao->byte_format != MACHINE_AO_FMT) { + *(*pp)++ = "-x"; + } + switch(ao->bits) { + case 8: *(*pp)++ = "-b"; break; + case 16: *(*pp)++ = "-w"; break; + case 32: *(*pp)++ = "-l"; break; + case 64: *(*pp)++ = "-d"; break; + default: fatal(0, "cannot handle sample size %d", (int)ao->bits); + } + break; + case 1: + switch(ao->byte_format) { + case AO_FMT_NATIVE: break; + case AO_FMT_BIG: *(*pp)++ = "-B"; break; + case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; + } + *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; + break; + } +} + +/** @brief Enable format translation + * + * If necessary, replaces a tracks inbound file descriptor with one connected + * to a sox invocation, which performs the required translation. + */ +static void enable_translation(struct track *t) { + switch(config->speaker_backend) { + case BACKEND_COMMAND: + case BACKEND_NETWORK: + /* These backends need a specific sample format */ + break; + case BACKEND_ALSA: + /* ALSA can cope */ + return; + } + if(!formats_equal(&t->format, &config->sample_format)) { + char argbuf[1024], *q = argbuf; + const char *av[18], **pp = av; + int soxpipe[2]; + pid_t soxkid; + + *pp++ = "sox"; + soxargs(&pp, &q, &t->format); + *pp++ = "-"; + soxargs(&pp, &q, &config->sample_format); + *pp++ = "-"; + *pp++ = 0; + if(debugging) { + for(pp = av; *pp; pp++) + D(("sox arg[%d] = %s", pp - av, *pp)); + D(("end args")); + } + xpipe(soxpipe); + soxkid = xfork(); + if(soxkid == 0) { + signal(SIGPIPE, SIG_DFL); + xdup2(t->fd, 0); + xdup2(soxpipe[1], 1); + fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); + close(soxpipe[0]); + close(soxpipe[1]); + close(t->fd); + execvp("sox", (char **)av); + _exit(1); + } + D(("forking sox for format conversion (kid = %d)", soxkid)); + close(t->fd); + close(soxpipe[1]); + t->fd = soxpipe[0]; + t->format = config->sample_format; + ready = 0; + } +} + +/** @brief Read data into a sample buffer + * @param t Pointer to track + * @return 0 on success, -1 on EOF + * + * This is effectively the read callback on @c t->fd. + */ static int fill(struct track *t) { size_t where, left; int n; @@ -242,6 +379,8 @@ static int fill(struct track *t) { /* Check that our assumptions are met. */ if(t->format.bits & 7) fatal(0, "bits per sample not a multiple of 8"); + /* If the input format is unsuitable, arrange to translate it */ + enable_translation(t); /* Make a new buffer for audio data. */ t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; t->buffer = xmalloc(t->size); @@ -253,16 +392,7 @@ static int fill(struct track *t) { return 0; } -/* Return true if A and B denote identical libao formats, else false. */ -static int formats_equal(const ao_sample_format *a, - const ao_sample_format *b) { - return (a->bits == b->bits - && a->rate == b->rate - && a->channels == b->channels - && a->byte_format == b->byte_format); -} - -/* Close the sound device. */ +/** @brief Close the sound device */ static void idle(void) { D(("idle")); #if API_ALSA @@ -284,7 +414,7 @@ static void idle(void) { ready = 0; } -/* Abandon the current track */ +/** @brief Abandon the current track */ static void abandon(void) { struct speaker_message sm; @@ -300,6 +430,7 @@ static void abandon(void) { } #if API_ALSA +/** @brief Log ALSA parameters */ static void log_params(snd_pcm_hw_params_t *hwparams, snd_pcm_sw_params_t *swparams) { snd_pcm_uframes_t f; @@ -326,43 +457,11 @@ static void log_params(snd_pcm_hw_params_t *hwparams, } #endif -static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { - int n; - - *(*pp)++ = "-t.raw"; - *(*pp)++ = "-s"; - *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; - *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; - /* sox 12.17.9 insists on -b etc; CVS sox insists on - etc; both are - * deployed! */ - switch(config->sox_generation) { - case 0: - if(ao->bits != 8 - && ao->byte_format != AO_FMT_NATIVE - && ao->byte_format != MACHINE_AO_FMT) { - *(*pp)++ = "-x"; - } - switch(ao->bits) { - case 8: *(*pp)++ = "-b"; break; - case 16: *(*pp)++ = "-w"; break; - case 32: *(*pp)++ = "-l"; break; - case 64: *(*pp)++ = "-d"; break; - default: fatal(0, "cannot handle sample size %d", (int)ao->bits); - } - break; - case 1: - switch(ao->byte_format) { - case AO_FMT_NATIVE: break; - case AO_FMT_BIG: *(*pp)++ = "-B"; break; - case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; - } - *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; - break; - } -} - -/* Make sure the sound device is open and has the right sample format. Return - * 0 on success and -1 on error. */ +/** @brief Enable sound output + * + * Makes sure the sound device is open and has the right sample format. Return + * 0 on success and -1 on error. + */ static int activate(void) { /* If we don't know the format yet we cannot start. */ if(!playing->got_format) { @@ -372,43 +471,6 @@ static int activate(void) { switch(config->speaker_backend) { case BACKEND_COMMAND: case BACKEND_NETWORK: - /* If we pass audio on to some other agent then we enforce the configured - * sample format on the *inbound* audio data. */ - if(!formats_equal(&playing->format, &config->sample_format)) { - char argbuf[1024], *q = argbuf; - const char *av[18], **pp = av; - int soxpipe[2]; - pid_t soxkid; - *pp++ = "sox"; - soxargs(&pp, &q, &playing->format); - *pp++ = "-"; - soxargs(&pp, &q, &config->sample_format); - *pp++ = "-"; - *pp++ = 0; - if(debugging) { - for(pp = av; *pp; pp++) - D(("sox arg[%d] = %s", pp - av, *pp)); - D(("end args")); - } - xpipe(soxpipe); - soxkid = xfork(); - if(soxkid == 0) { - xdup2(playing->fd, 0); - xdup2(soxpipe[1], 1); - fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); - close(soxpipe[0]); - close(soxpipe[1]); - close(playing->fd); - execvp("sox", (char **)av); - _exit(1); - } - D(("forking sox for format conversion (kid = %d)", soxkid)); - close(playing->fd); - close(soxpipe[1]); - playing->fd = soxpipe[0]; - playing->format = config->sample_format; - ready = 0; - } if(!ready) { pcm_format = config->sample_format; bufsize = 3 * FRAMES; @@ -541,6 +603,7 @@ static void fork_cmd(void) { xpipe(pfd); cmdpid = xfork(); if(!cmdpid) { + signal(SIGPIPE, SIG_DFL); xdup2(pfd[0], 0); close(pfd[0]); close(pfd[1]); -- [mdw]