X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/09ee2f0d809da23b6b442233b5c3d94a6e64bdd2..5aff007d8fcfb4c6cc3c3627ae15f45562db7a0d:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index 5606fb5..31416ec 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -1,6 +1,6 @@ /* * This file is part of DisOrder. - * Copyright (C) 2007 Richard Kettlewell + * Copyright (C) 2007, 2008 Richard Kettlewell * * 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 @@ -17,6 +17,37 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 * USA */ +/** @file clients/playrtp.c + * @brief RTP player + * + * This player supports Linux (ALSA) + * and Apple Mac (Core Audio) + * systems. There is no support for Microsoft Windows yet, and that will in + * fact probably an entirely separate program. + * + * The program runs (at least) three threads. listen_thread() is responsible + * for reading RTP packets off the wire and adding them to the linked list @ref + * received_packets, assuming they are basically sound. queue_thread() takes + * packets off this linked list and adds them to @ref packets (an operation + * which might be much slower due to contention for @ref lock). + * + * The main thread is responsible for actually playing audio. In ALSA this + * means it waits until ALSA says it's ready for more audio which it then + * plays. See @ref clients/playrtp-alsa.c. + * + * In Core Audio the main thread is only responsible for starting and stopping + * play: the system does the actual playback in its own private thread, and + * calls adioproc() to fetch the audio data. See @ref + * clients/playrtp-coreaudio.c. + * + * Sometimes it happens that there is no audio available to play. This may + * because the server went away, or a packet was dropped, or the server + * deliberately did not send any sound because it encountered a silence. + * + * Assumptions: + * - it is safe to read uint32_t values without a lock protecting them + */ #include #include "types.h" @@ -30,6 +61,16 @@ #include #include #include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include #include "log.h" #include "mem.h" @@ -38,122 +79,312 @@ #include "syscalls.h" #include "rtp.h" #include "defs.h" - -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -# include -#endif -#if API_ALSA -#include +#include "vector.h" +#include "heap.h" +#include "timeval.h" +#include "client.h" +#include "playrtp.h" +#include "inputline.h" +#include "version.h" + +#define readahead linux_headers_are_borked + +/** @brief Obsolete synonym */ +#ifndef IPV6_JOIN_GROUP +# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP #endif /** @brief RTP socket */ static int rtpfd; +/** @brief Log output */ +static FILE *logfp; + /** @brief Output device */ -static const char *device; +const char *device; -/** @brief Maximum samples per packet we'll support +/** @brief Minimum low watermark * - * NB that two channels = two samples in this program. - */ -#define MAXSAMPLES 2048 + * We'll stop playing if there's only this many samples in the buffer. */ +unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ -/** @brief Minimum buffer size +/** @brief Buffer high watermark * - * We'll stop playing if there's only this many samples in the buffer. */ -#define MINBUFFER 8820 + * We'll only start playing when this many samples are available. */ +static unsigned readahead = 2 * 2 * 44100; -/** @brief Maximum sample size +/** @brief Maximum buffer size * - * The maximum supported size (in bytes) of one sample. */ -#define MAXSAMPLESIZE 2 + * We'll stop reading from the network if we have this many samples. */ +static unsigned maxbuffer; -#define READAHEAD 88200 /* how far to read ahead */ +/** @brief Received packets + * Protected by @ref receive_lock + * + * Received packets are added to this list, and queue_thread() picks them off + * it and adds them to @ref packets. Whenever a packet is added to it, @ref + * receive_cond is signalled. + */ +struct packet *received_packets; -#define MAXBUFFER (3 * 88200) /* maximum buffer contents */ +/** @brief Tail of @ref received_packets + * Protected by @ref receive_lock + */ +struct packet **received_tail = &received_packets; -/** @brief Received packet +/** @brief Lock protecting @ref received_packets * - * Packets are recorded in an ordered linked list. */ -struct packet { - /** @brief Pointer to next packet - * The next packet might not be immediately next: if packets are dropped - * or mis-ordered there may be gaps at any given moment. */ - struct packet *next; - /** @brief Number of samples in this packet */ - int nsamples; - /** @brief Number of samples used from this packet */ - int nused; - /** @brief Timestamp from RTP packet - * - * NB that "timestamps" are really sample counters.*/ - uint32_t timestamp; -#if HAVE_COREAUDIO_AUDIOHARDWARE_H - /** @brief Converted sample data */ - float samples_float[MAXSAMPLES]; -#else - /** @brief Raw sample data */ - unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE]; -#endif -}; + * Only listen_thread() and queue_thread() ever hold this lock. It is vital + * that queue_thread() not hold it any longer than it strictly has to. */ +pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER; + +/** @brief Condition variable signalled when @ref received_packets is updated + * + * Used by listen_thread() to notify queue_thread() that it has added another + * packet to @ref received_packets. */ +pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER; -/** @brief Total number of samples available */ -static unsigned long nsamples; +/** @brief Length of @ref received_packets */ +uint32_t nreceived; -/** @brief Linked list of packets +/** @brief Binary heap of received packets */ +struct pheap packets; + +/** @brief Total number of samples available * - * In ascending order of timestamp. */ -static struct packet *packets; + * We make this volatile because we inspect it without a protecting lock, + * so the usual pthread_* guarantees aren't available. + */ +volatile uint32_t nsamples; /** @brief Timestamp of next packet to play. * * This is set to the timestamp of the last packet, plus the number of * samples it contained. Only valid if @ref active is nonzero. */ -static uint32_t next_timestamp; +uint32_t next_timestamp; /** @brief True if actively playing * * This is true when playing and false when just buffering. */ -static int active; +int active; /** @brief Lock protecting @ref packets */ -static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; +pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; /** @brief Condition variable signalled whenever @ref packets is changed */ -static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; +pthread_cond_t cond = PTHREAD_COND_INITIALIZER; + +#if HAVE_ALSA_ASOUNDLIB_H +# define DEFAULT_BACKEND playrtp_alsa +#elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST +# define DEFAULT_BACKEND playrtp_oss +#elif HAVE_COREAUDIO_AUDIOHARDWARE_H +# define DEFAULT_BACKEND playrtp_coreaudio +#else +# error No known backend +#endif + +/** @brief Backend to play with */ +static void (*backend)(void) = &DEFAULT_BACKEND; + +HEAP_DEFINE(pheap, struct packet *, lt_packet); + +/** @brief Control socket or NULL */ +const char *control_socket; + +/** @brief Buffer for debugging dump + * + * The debug dump is enabled by the @c --dump option. It records the last 20s + * of audio to the specified file (which will be about 3.5Mbytes). The file is + * written as as ring buffer, so the start point will progress through it. + * + * Use clients/dump2wav to convert this to a WAV file, which can then be loaded + * into (e.g.) Audacity for further inspection. + * + * All three backends (ALSA, OSS, Core Audio) now support this option. + * + * The idea is to allow the user a few seconds to react to an audible artefact. + */ +int16_t *dump_buffer; + +/** @brief Current index within debugging dump */ +size_t dump_index; + +/** @brief Size of debugging dump in samples */ +size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/; static const struct option options[] = { { "help", no_argument, 0, 'h' }, { "version", no_argument, 0, 'V' }, { "debug", no_argument, 0, 'd' }, { "device", required_argument, 0, 'D' }, + { "min", required_argument, 0, 'm' }, + { "max", required_argument, 0, 'x' }, + { "buffer", required_argument, 0, 'b' }, + { "rcvbuf", required_argument, 0, 'R' }, +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + { "oss", no_argument, 0, 'o' }, +#endif +#if HAVE_ALSA_ASOUNDLIB_H + { "alsa", no_argument, 0, 'a' }, +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + { "core-audio", no_argument, 0, 'c' }, +#endif + { "dump", required_argument, 0, 'r' }, + { "socket", required_argument, 0, 's' }, + { "config", required_argument, 0, 'C' }, { 0, 0, 0, 0 } }; -/** @brief Return true iff a < b in sequence-space arithmetic */ -static inline int lt(uint32_t a, uint32_t b) { - return (uint32_t)(a - b) & 0x80000000; +/** @brief Control thread + * + * This thread is responsible for accepting control commands from Disobedience + * (or other controllers) over an AF_UNIX stream socket with a path specified + * by the @c --socket option. The protocol uses simple string commands and + * replies: + * + * - @c stop will shut the player down + * - @c query will send back the reply @c running + * - anything else is ignored + * + * Commands and response strings terminated by shutting down the connection or + * by a newline. No attempt is made to multiplex multiple clients so it is + * important that the command be sent as soon as the connection is made - it is + * assumed that both parties to the protocol are entirely cooperating with one + * another. + */ +static void *control_thread(void attribute((unused)) *arg) { + struct sockaddr_un sa; + int sfd, cfd; + char *line; + socklen_t salen; + FILE *fp; + + assert(control_socket); + unlink(control_socket); + memset(&sa, 0, sizeof sa); + sa.sun_family = AF_UNIX; + strcpy(sa.sun_path, control_socket); + sfd = xsocket(PF_UNIX, SOCK_STREAM, 0); + if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0) + fatal(errno, "error binding to %s", control_socket); + if(listen(sfd, 128) < 0) + fatal(errno, "error calling listen on %s", control_socket); + info("listening on %s", control_socket); + for(;;) { + salen = sizeof sa; + cfd = accept(sfd, (struct sockaddr *)&sa, &salen); + if(cfd < 0) { + switch(errno) { + case EINTR: + case EAGAIN: + break; + default: + fatal(errno, "error calling accept on %s", control_socket); + } + } + if(!(fp = fdopen(cfd, "r+"))) { + error(errno, "error calling fdopen for %s connection", control_socket); + close(cfd); + continue; + } + if(!inputline(control_socket, fp, &line, '\n')) { + if(!strcmp(line, "stop")) { + info("stopped via %s", control_socket); + exit(0); /* terminate immediately */ + } + if(!strcmp(line, "query")) + fprintf(fp, "running"); + xfree(line); + } + if(fclose(fp) < 0) + error(errno, "error closing %s connection", control_socket); + } +} + +/** @brief Drop the first packet + * + * Assumes that @ref lock is held. + */ +static void drop_first_packet(void) { + if(pheap_count(&packets)) { + struct packet *const p = pheap_remove(&packets); + nsamples -= p->nsamples; + playrtp_free_packet(p); + pthread_cond_broadcast(&cond); + } +} + +/** @brief Background thread adding packets to heap + * + * This just transfers packets from @ref received_packets to @ref packets. It + * is important that it holds @ref receive_lock for as little time as possible, + * in order to minimize the interval between calls to read() in + * listen_thread(). + */ +static void *queue_thread(void attribute((unused)) *arg) { + struct packet *p; + + for(;;) { + /* Get the next packet */ + pthread_mutex_lock(&receive_lock); + while(!received_packets) { + pthread_cond_wait(&receive_cond, &receive_lock); + } + p = received_packets; + received_packets = p->next; + if(!received_packets) + received_tail = &received_packets; + --nreceived; + pthread_mutex_unlock(&receive_lock); + /* Add it to the heap */ + pthread_mutex_lock(&lock); + pheap_insert(&packets, p); + nsamples += p->nsamples; + pthread_cond_broadcast(&cond); + pthread_mutex_unlock(&lock); + } } /** @brief Background thread collecting samples * * This function collects samples, perhaps converts them to the target format, - * and adds them to the packet list. */ + * and adds them to the packet list. + * + * It is crucial that the gap between successive calls to read() is as small as + * possible: otherwise packets will be dropped. + * + * We use a binary heap to ensure that the unavoidable effort is at worst + * logarithmic in the total number of packets - in fact if packets are mostly + * received in order then we will largely do constant work per packet since the + * newest packet will always be last. + * + * Of more concern is that we must acquire the lock on the heap to add a packet + * to it. If this proves a problem in practice then the answer would be + * (probably doubly) linked list with new packets added the end and a second + * thread which reads packets off the list and adds them to the heap. + * + * We keep memory allocation (mostly) very fast by keeping pre-allocated + * packets around; see @ref playrtp_new_packet(). + */ static void *listen_thread(void attribute((unused)) *arg) { - struct packet *p = 0, **pp; + struct packet *p = 0; int n; - union { - struct rtp_header header; - uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)]; - } packet; - const uint16_t *const samples = (uint16_t *)(packet.bytes - + sizeof (struct rtp_header)); + struct rtp_header header; + uint16_t seq; + uint32_t timestamp; + struct iovec iov[2]; for(;;) { if(!p) - p = xmalloc(sizeof *p); - n = read(rtpfd, packet.bytes, sizeof packet.bytes); + p = playrtp_new_packet(); + iov[0].iov_base = &header; + iov[0].iov_len = sizeof header; + iov[1].iov_base = p->samples_raw; + iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; + n = readv(rtpfd, iov, 2); if(n < 0) { switch(errno) { case EINTR: @@ -163,107 +394,96 @@ static void *listen_thread(void attribute((unused)) *arg) { } } /* Ignore too-short packets */ - if((size_t)n <= sizeof (struct rtp_header)) + if((size_t)n <= sizeof (struct rtp_header)) { + info("ignored a short packet"); continue; - p->nused = 0; - p->timestamp = ntohl(packet.header.timestamp); + } + timestamp = htonl(header.timestamp); + seq = htons(header.seq); /* Ignore packets in the past */ - if(active && lt(p->timestamp, next_timestamp)) + if(active && lt(timestamp, next_timestamp)) { + info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, + timestamp, next_timestamp); continue; + } + p->next = 0; + p->flags = 0; + p->timestamp = timestamp; /* Convert to target format */ - switch(packet.header.mpt & 0x7F) { + if(header.mpt & 0x80) + p->flags |= IDLE; + switch(header.mpt & 0x7F) { case 10: - p->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t); -#if HAVE_COREAUDIO_AUDIOHARDWARE_H - /* Convert to what Core Audio expects */ - for(n = 0; n < p->nsamples; ++n) - p->samples_float[n] = (int16_t)ntohs(samples[n]) * (0.5f / 32767); -#else - /* ALSA can do any necessary conversion itself (though it might be better - * to do any necessary conversion in the background) */ - memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header)); -#endif + p->nsamples = (n - sizeof header) / sizeof(uint16_t); break; /* TODO support other RFC3551 media types (when the speaker does) */ default: fatal(0, "unsupported RTP payload type %d", - packet.header.mpt & 0x7F); + header.mpt & 0x7F); } - pthread_mutex_lock(&lock); + if(logfp) + fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", + seq, timestamp, p->nsamples, timestamp + p->nsamples); /* Stop reading if we've reached the maximum. * * This is rather unsatisfactory: it means that if packets get heavily * out of order then we guarantee dropouts. But for now... */ - while(nsamples >= MAXBUFFER) - pthread_cond_wait(&cond, &lock); - for(pp = &packets; - *pp && lt((*pp)->timestamp, p->timestamp); - pp = &(*pp)->next) - ; - /* So now either !*pp or *pp >= p */ - if(*pp && p->timestamp == (*pp)->timestamp) { - /* *pp == p; a duplicate. Ideally we avoid the translation step here, - * but we'll worry about that another time. */ - } else { - p->next = *pp; - *pp = p; - nsamples += p->nsamples; - pthread_cond_broadcast(&cond); - p = 0; /* we've consumed this packet */ + if(nsamples >= maxbuffer) { + pthread_mutex_lock(&lock); + while(nsamples >= maxbuffer) { + pthread_cond_wait(&cond, &lock); + } + pthread_mutex_unlock(&lock); } - pthread_mutex_unlock(&lock); + /* Add the packet to the receive queue */ + pthread_mutex_lock(&receive_lock); + *received_tail = p; + received_tail = &p->next; + ++nreceived; + pthread_cond_signal(&receive_cond); + pthread_mutex_unlock(&receive_lock); + /* We'll need a new packet */ + p = 0; } } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -/** @brief Callback from Core Audio */ -static OSStatus adioproc(AudioDeviceID inDevice, - const AudioTimeStamp *inNow, - const AudioBufferList *inInputData, - const AudioTimeStamp *inInputTime, - AudioBufferList *outOutputData, - const AudioTimeStamp *inOutputTime, - void *inClientData) { - UInt32 nbuffers = outOutputData->mNumberBuffers; - AudioBuffer *ab = outOutputData->mBuffers; - float *samplesOut; /* where to write samples to */ - size_t samplesOutLeft; /* space left */ - size_t samplesInLeft; - size_t samplesToCopy; - - pthread_mutex_lock(&lock); - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); - while(packets && nbuffers > 0) { - if(packets->used == packets->nsamples) { - /* TODO if we dropped a packet then we should introduce a gap here */ - struct packet *const p = packets; - packets = p->next; - free(p); - pthread_cond_broadcast(&cond); - continue; - } - if(samplesOutLeft == 0) { - --nbuffers; - ++ab; - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); - continue; - } - /* Now: (1) there is some data left to read - * (2) there is some space to put it */ - samplesInLeft = packets->nsamples - packets->used; - samplesToCopy = (samplesInLeft < samplesOutLeft - ? samplesInLeft : samplesOutLeft); - memcpy(samplesOut, packet->samples + packets->used, samplesToCopy); - packets->used += samplesToCopy; - samplesOut += samplesToCopy; - samesOutLeft -= samplesToCopy; +/** @brief Wait until the buffer is adequately full + * + * Must be called with @ref lock held. + */ +void playrtp_fill_buffer(void) { + while(nsamples) + drop_first_packet(); + info("Buffering..."); + while(nsamples < readahead) { + pthread_cond_wait(&cond, &lock); + } + next_timestamp = pheap_first(&packets)->timestamp; + active = 1; +} + +/** @brief Find next packet + * @return Packet to play or NULL if none found + * + * The return packet is merely guaranteed not to be in the past: it might be + * the first packet in the future rather than one that is actually suitable to + * play. + * + * Must be called with @ref lock held. + */ +struct packet *playrtp_next_packet(void) { + while(pheap_count(&packets)) { + struct packet *const p = pheap_first(&packets); + if(le(p->timestamp + p->nsamples, next_timestamp)) { + /* This packet is in the past. Drop it and try another one. */ + drop_first_packet(); + } else + /* This packet is NOT in the past. (It might be in the future + * however.) */ + return p; } - pthread_mutex_unlock(&lock); return 0; } -#endif /** @brief Play an RTP stream * @@ -276,196 +496,16 @@ static OSStatus adioproc(AudioDeviceID inDevice, */ static void play_rtp(void) { pthread_t ltid; + int err; /* We receive and convert audio data in a background thread */ - pthread_create(<id, 0, listen_thread, 0); -#if API_ALSA - { - snd_pcm_t *pcm; - snd_pcm_hw_params_t *hwparams; - snd_pcm_sw_params_t *swparams; - /* Only support one format for now */ - const int sample_format = SND_PCM_FORMAT_S16_BE; - unsigned rate = 44100; - const int channels = 2; - const int samplesize = channels * sizeof(uint16_t); - snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; - /* If we can write more than this many samples we'll get a wakeup */ - const int avail_min = 256; - snd_pcm_sframes_t frames_written; - size_t samples_written; - int prepared = 1; - int err; - - /* Open ALSA */ - if((err = snd_pcm_open(&pcm, - device ? device : "default", - SND_PCM_STREAM_PLAYBACK, - SND_PCM_NONBLOCK))) - fatal(0, "error from snd_pcm_open: %d", err); - /* Set up 'hardware' parameters */ - snd_pcm_hw_params_alloca(&hwparams); - if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) - fatal(0, "error from snd_pcm_hw_params_any: %d", err); - if((err = snd_pcm_hw_params_set_access(pcm, hwparams, - SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); - if((err = snd_pcm_hw_params_set_format(pcm, hwparams, - sample_format)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", - sample_format, err); - if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", - rate, err); - if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, - channels)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", - channels, err); - if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, - &pcm_bufsize)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", - MAXSAMPLES * samplesize * 3, err); - if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) - fatal(0, "error calling snd_pcm_hw_params: %d", err); - /* Set up 'software' parameters */ - snd_pcm_sw_params_alloca(&swparams); - if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) - fatal(0, "error calling snd_pcm_sw_params_current: %d", err); - if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) - fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", - avail_min, err); - if((err = snd_pcm_sw_params(pcm, swparams)) < 0) - fatal(0, "error calling snd_pcm_sw_params: %d", err); - - /* Ready to go */ - - pthread_mutex_lock(&lock); - for(;;) { - /* Wait for the buffer to fill up a bit */ - while(nsamples < READAHEAD) - pthread_cond_wait(&cond, &lock); - if(!prepared) { - if((err = snd_pcm_prepare(pcm))) - fatal(0, "error calling snd_pcm_prepare: %d", err); - prepared = 1; - } - /* Start at the first available packet */ - next_timestamp = packets->timestamp; - active = 1; - /* Wait until the buffer empties out */ - while(nsamples >= MINBUFFER) { - /* Wait for ALSA to ask us for more data */ - pthread_mutex_unlock(&lock); - snd_pcm_wait(pcm, -1); - pthread_mutex_lock(&lock); - /* ALSA is ready for more data */ - if(packets && packets->timestamp + packets->nused == next_timestamp) { - /* Hooray, we have a packet we can play */ - const size_t samples_available = packets->nsamples - packets->nused; - const size_t frames_available = samples_available / 2; - - frames_written = snd_pcm_writei(pcm, - packets->samples_raw + packets->nused, - frames_available); - if(frames_written < 0) - fatal(0, "error calling snd_pcm_writei: %d", err); - samples_written = frames_written * 2; - packets->nused += samples_written; - next_timestamp += samples_written; - if(packets->nused == packets->nsamples) { - /* We're done with this packet */ - struct packet *p = packets; - - packets = p->next; - nsamples -= p->nsamples; - free(p); - pthread_cond_broadcast(&cond); - } - } else { - /* We don't have anything to play! We'd better play some 0s. */ - static const uint16_t zeros[1024]; - size_t samples_available = 1024, frames_available; - if(packets && next_timestamp + samples_available > packets->timestamp) - samples_available = packets->timestamp - next_timestamp; - frames_available = samples_available / 2; - frames_written = snd_pcm_writei(pcm, - zeros, - frames_available); - if(frames_written < 0) - fatal(0, "error calling snd_pcm_writei: %d", err); - next_timestamp += samples_written; - } - } - active = 0; - /* We stop playing for a bit until the buffer re-fills */ - pthread_mutex_unlock(&lock); - if((err = snd_pcm_drain(pcm))) - fatal(0, "error calling snd_pcm_drain: %d", err); - prepared = 0; - pthread_mutex_lock(&lock); - } - - } -#elif HAVE_COREAUDIO_AUDIOHARDWARE_H - { - OSStatus status; - UInt32 propertySize; - AudioDeviceID adid; - AudioStreamBasicDescription asbd; - - /* If this looks suspiciously like libao's macosx driver there's an - * excellent reason for that... */ - - /* TODO report errors as strings not numbers */ - propertySize = sizeof adid; - status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, - &propertySize, &adid); - if(status) - fatal(0, "AudioHardwareGetProperty: %d", (int)status); - if(adid == kAudioDeviceUnknown) - fatal(0, "no output device"); - propertySize = sizeof asbd; - status = AudioDeviceGetProperty(adid, 0, false, - kAudioDevicePropertyStreamFormat, - &propertySize, &asbd); - if(status) - fatal(0, "AudioHardwareGetProperty: %d", (int)status); - D(("mSampleRate %f", asbd.mSampleRate)); - D(("mFormatID %08"PRIx32, asbd.mFormatID)); - D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags)); - D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket)); - D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket)); - D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame)); - D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame)); - D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel)); - D(("mReserved %08"PRIx32, asbd.mReserved)); - if(asbd.mFormatID != kAudioFormatLinearPCM) - fatal(0, "audio device does not support kAudioFormatLinearPCM"); - status = AudioDeviceAddIOProc(adid, adioproc, 0); - if(status) - fatal(0, "AudioDeviceAddIOProc: %d", (int)status); - pthread_mutex_lock(&lock); - for(;;) { - /* Wait for the buffer to fill up a bit */ - while(nsamples < READAHEAD) - pthread_cond_wait(&cond, &lock); - /* Start playing now */ - status = AudioDeviceStart(adid, adioproc); - if(status) - fatal(0, "AudioDeviceStart: %d", (int)status); - /* Wait until the buffer empties out */ - while(nsamples >= MINBUFFER) - pthread_cond_wait(&cond, &lock); - /* Stop playing for a bit until the buffer re-fills */ - status = AudioDeviceStop(adid, adioproc); - if(status) - fatal(0, "AudioDeviceStop: %d", (int)status); - /* Go back round */ - } - } -#else -# error No known audio API -#endif + if((err = pthread_create(<id, 0, listen_thread, 0))) + fatal(err, "pthread_create listen_thread"); + /* We have a second thread to add received packets to the queue */ + if((err = pthread_create(<id, 0, queue_thread, 0))) + fatal(err, "pthread_create queue_thread"); + /* The rest of the work is backend-specific */ + backend(); } /* display usage message and terminate */ @@ -473,26 +513,47 @@ static void help(void) { xprintf("Usage:\n" " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" "Options:\n" + " --device, -D DEVICE Output device\n" + " --min, -m FRAMES Buffer low water mark\n" + " --buffer, -b FRAMES Buffer high water mark\n" + " --max, -x FRAMES Buffer maximum size\n" + " --rcvbuf, -R BYTES Socket receive buffer size\n" + " --config, -C PATH Set configuration file\n" +#if HAVE_ALSA_ASOUNDLIB_H + " --alsa, -a Use ALSA to play audio\n" +#endif +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + " --oss, -o Use OSS to play audio\n" +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + " --core-audio, -c Use Core Audio to play audio\n" +#endif " --help, -h Display usage message\n" " --version, -V Display version number\n" - " --debug, -d Turn on debugging\n" - " --device, -D DEVICE Output device\n"); - xfclose(stdout); - exit(0); -} - -/* display version number and terminate */ -static void version(void) { - xprintf("disorder-playrtp version %s\n", disorder_version_string); + ); xfclose(stdout); exit(0); } int main(int argc, char **argv) { - int n; + int n, err; struct addrinfo *res; struct stringlist sl; char *sockname; + int rcvbuf, target_rcvbuf = 131072; + socklen_t len; + struct ip_mreq mreq; + struct ipv6_mreq mreq6; + disorder_client *c; + char *address, *port; + int is_multicast; + union any_sockaddr { + struct sockaddr sa; + struct sockaddr_in in; + struct sockaddr_in6 in6; + }; + union any_sockaddr mgroup; + const char *dumpfile = 0; static const struct addrinfo prefs = { AI_PASSIVE, @@ -507,30 +568,158 @@ int main(int argc, char **argv) { mem_init(); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVdD", options, 0)) >= 0) { + while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) { switch(n) { case 'h': help(); - case 'V': version(); + case 'V': version("disorder-playrtp"); case 'd': debugging = 1; break; case 'D': device = optarg; break; + case 'm': minbuffer = 2 * atol(optarg); break; + case 'b': readahead = 2 * atol(optarg); break; + case 'x': maxbuffer = 2 * atol(optarg); break; + case 'L': logfp = fopen(optarg, "w"); break; + case 'R': target_rcvbuf = atoi(optarg); break; +#if HAVE_ALSA_ASOUNDLIB_H + case 'a': backend = playrtp_alsa; break; +#endif +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + case 'o': backend = playrtp_oss; break; +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + case 'c': backend = playrtp_coreaudio; break; +#endif + case 'C': configfile = optarg; break; + case 's': control_socket = optarg; break; + case 'r': dumpfile = optarg; break; default: fatal(0, "invalid option"); } } + if(config_read(0)) fatal(0, "cannot read configuration"); + if(!maxbuffer) + maxbuffer = 4 * readahead; argc -= optind; argv += optind; - if(argc < 1 || argc > 2) - fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); - sl.n = argc; - sl.s = argv; - /* Listen for inbound audio data */ + switch(argc) { + case 0: + /* Get configuration from server */ + if(!(c = disorder_new(1))) exit(EXIT_FAILURE); + if(disorder_connect(c)) exit(EXIT_FAILURE); + if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE); + sl.n = 2; + sl.s = xcalloc(2, sizeof *sl.s); + sl.s[0] = address; + sl.s[1] = port; + break; + case 1: + case 2: + /* Use command-line ADDRESS+PORT or just PORT */ + sl.n = argc; + sl.s = argv; + break; + default: + fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]"); + } + /* Look up address and port */ if(!(res = get_address(&sl, &prefs, &sockname))) exit(1); + /* Create the socket */ if((rtpfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol)) < 0) fatal(errno, "error creating socket"); + /* Stash the multicast group address */ + if((is_multicast = multicast(res->ai_addr))) { + memcpy(&mgroup, res->ai_addr, res->ai_addrlen); + switch(res->ai_addr->sa_family) { + case AF_INET: + mgroup.in.sin_port = 0; + break; + case AF_INET6: + mgroup.in6.sin6_port = 0; + break; + } + } + /* Bind to 0/port */ + switch(res->ai_addr->sa_family) { + case AF_INET: + memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0, + sizeof (struct in_addr)); + break; + case AF_INET6: + memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0, + sizeof (struct in6_addr)); + break; + default: + fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family); + } if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) fatal(errno, "error binding socket to %s", sockname); + if(is_multicast) { + switch(mgroup.sa.sa_family) { + case PF_INET: + mreq.imr_multiaddr = mgroup.in.sin_addr; + mreq.imr_interface.s_addr = 0; /* use primary interface */ + if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP, + &mreq, sizeof mreq) < 0) + fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP"); + break; + case PF_INET6: + mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr; + memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface); + if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP, + &mreq6, sizeof mreq6) < 0) + fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); + break; + default: + fatal(0, "unsupported address family %d", res->ai_family); + } + info("listening on %s multicast group %s", + format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa)); + } else + info("listening on %s", format_sockaddr(res->ai_addr)); + len = sizeof rcvbuf; + if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) + fatal(errno, "error calling getsockopt SO_RCVBUF"); + if(target_rcvbuf > rcvbuf) { + if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, + &target_rcvbuf, sizeof target_rcvbuf) < 0) + error(errno, "error calling setsockopt SO_RCVBUF %d", + target_rcvbuf); + /* We try to carry on anyway */ + else + info("changed socket receive buffer from %d to %d", + rcvbuf, target_rcvbuf); + } else + info("default socket receive buffer %d", rcvbuf); + if(logfp) + info("WARNING: -L option can impact performance"); + if(control_socket) { + pthread_t tid; + + if((err = pthread_create(&tid, 0, control_thread, 0))) + fatal(err, "pthread_create control_thread"); + } + if(dumpfile) { + int fd; + unsigned char buffer[65536]; + size_t written; + + if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0) + fatal(errno, "opening %s", dumpfile); + /* Fill with 0s to a suitable size */ + memset(buffer, 0, sizeof buffer); + for(written = 0; written < dump_size * sizeof(int16_t); + written += sizeof buffer) { + if(write(fd, buffer, sizeof buffer) < 0) + fatal(errno, "clearing %s", dumpfile); + } + /* Map the buffer into memory for convenience */ + dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE, + MAP_SHARED, fd, 0); + if(dump_buffer == (void *)-1) + fatal(errno, "mapping %s", dumpfile); + info("dumping to %s", dumpfile); + } play_rtp(); return 0; }