X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/e83d0967d4c0965eb8036248acc20d1bf12ad1d8..cca034e58b00be2f53f9145957346bcd3e451c45:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index 7ec35d0..89e458e 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -1,34 +1,73 @@ /* * This file is part of DisOrder. - * Copyright (C) 2007 Richard Kettlewell + * Copyright (C) 2007-2009, 2011, 2013 Richard Kettlewell * - * This program is free software; you can redistribute it and/or modify + * 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 + * the Free Software Foundation, either version 3 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. - * + * 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, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 - * USA + * along with this program. If not, see . + */ +/** @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). + * + * control_thread() accepts commands from Disobedience (or anything else). + * + * The main thread activates and deactivates audio playing via the @ref + * lib/uaudio.h API (which probably implies at least one further thread). + * + * 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" +#include "common.h" #include -#include -#include #include #include #include #include #include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include #include "log.h" #include "mem.h" @@ -36,287 +75,944 @@ #include "addr.h" #include "syscalls.h" #include "rtp.h" -#include "debug.h" +#include "defs.h" +#include "vector.h" +#include "heap.h" +#include "timeval.h" +#include "client.h" +#include "playrtp.h" +#include "inputline.h" +#include "version.h" +#include "uaudio.h" -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -# include +/** @brief Obsolete synonym */ +#ifndef IPV6_JOIN_GROUP +# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP #endif +/** @brief RTP socket */ static int rtpfd; -#define MAXSAMPLES 2048 /* max samples/frame we'll support */ -/* NB two channels = two samples in this program! */ -#define MINBUFFER 8820 /* when to stop playing */ -#define READAHEAD 88200 /* how far to read ahead */ -#define MAXBUFFER (3 * 88200) /* maximum buffer contents */ - -struct frame { - struct frame *next; /* another frame */ - int nsamples; /* number of samples */ - int nused; /* number of samples used so far */ - uint32_t timestamp; /* timestamp from packet */ -#if HAVE_COREAUDIO_AUDIOHARDWARE_H - float samples[MAXSAMPLES]; /* converted sample data */ -#endif -}; +/** @brief Log output */ +static FILE *logfp; + +/** @brief Output device */ + +/** @brief Buffer low watermark in samples */ +unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */ + +/** @brief Maximum buffer size in samples + * + * We'll stop reading from the network if we have this many samples. + */ +static unsigned maxbuffer; -static unsigned long nsamples; /* total samples available */ +/** @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; + +/** @brief Tail of @ref received_packets + * Protected by @ref receive_lock + */ +struct packet **received_tail = &received_packets; -static struct frame *frames; /* received frames in ascending order - * of timestamp */ -static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; -/* lock protecting frame list */ +/** @brief Lock protecting @ref received_packets + * + * 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; -static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER; -/* signalled whenever we add a new frame */ +/** @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 Length of @ref received_packets */ +uint32_t nreceived; + +/** @brief Binary heap of received packets */ +struct pheap packets; + +/** @brief Total number of samples available + * + * 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. + */ +uint32_t next_timestamp; + +/** @brief True if actively playing + * + * This is true when playing and false when just buffering. */ +int active; + +/** @brief Lock protecting @ref packets */ +pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; + +/** @brief Condition variable signalled whenever @ref packets is changed */ +pthread_cond_t cond = PTHREAD_COND_INITIALIZER; + +/** @brief Backend to play with */ +static const struct uaudio *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' }, + { "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 + { "api", required_argument, 0, 'A' }, + { "dump", required_argument, 0, 'r' }, + { "command", required_argument, 0, 'e' }, + { "pause-mode", required_argument, 0, 'P' }, + { "socket", required_argument, 0, 's' }, + { "config", required_argument, 0, 'C' }, + { "monitor", no_argument, 0, 'M' }, { 0, 0, 0, 0 } }; -/* Return true iff a > b in sequence-space arithmetic */ -static inline int gt(const struct frame *a, const struct frame *b) { - return (uint32_t)(a->timestamp - b->timestamp) < 0x80000000; -} - -/* Background thread that reads frames over the network and add them to the - * list */ -static listen_thread(void attribute((unused)) *arg) { - struct frame *f = 0, **ff; - int n, i; - 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)); +/** @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) + disorder_fatal(errno, "error binding to %s", control_socket); + if(listen(sfd, 128) < 0) + disorder_fatal(errno, "error calling listen on %s", control_socket); + disorder_info("listening on %s", control_socket); for(;;) { - if(!f) - f = xmalloc(sizeof *f); - n = read(rtpfd, packet.bytes, sizeof packet.bytes); - if(n < 0) { + salen = sizeof sa; + cfd = accept(sfd, (struct sockaddr *)&sa, &salen); + if(cfd < 0) { switch(errno) { case EINTR: - continue; + case EAGAIN: + break; default: - fatal(errno, "error reading from socket"); + disorder_fatal(errno, "error calling accept on %s", control_socket); } } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H - /* Convert to target format */ - switch(packet.header.mtp & 0x7F) { - case 10: - f->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t); - for(i = 0; i < f->nsamples; ++i) - f->samples[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767); - break; - /* TODO support other RFC3551 media types (when the speaker does) */ - default: - fatal(0, "unsupported RTP payload type %d", - packet.header.mpt & 0x7F); + if(!(fp = fdopen(cfd, "r+"))) { + disorder_error(errno, "error calling fdopen for %s connection", control_socket); + close(cfd); + continue; } -#endif - f->used = 0; - f->timestamp = ntohl(packet.header.timestamp); + if(!inputline(control_socket, fp, &line, '\n')) { + if(!strcmp(line, "stop")) { + disorder_info("stopped via %s", control_socket); + exit(0); /* terminate immediately */ + } + if(!strcmp(line, "query")) + fprintf(fp, "running"); + xfree(line); + } + if(fclose(fp) < 0) + disorder_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); - /* Stop reading if we've reached the maximum */ - while(nsamples >= MAXBUFFER) - pthread_cond_wait(&cond, &lock); - for(ff = &frames; *ff && !gt(*ff, f); ff = &(*ff)->next) - ; - f->next = *ff; - *ff = f; - nsamples += f->nsamples; + pheap_insert(&packets, p); + nsamples += p->nsamples; pthread_cond_broadcast(&cond); pthread_mutex_unlock(&lock); - f = 0; } +#if HAVE_STUPID_GCC44 + return NULL; +#endif } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -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(frames && nbuffers > 0) { - if(frames->used == frames->nsamples) { - /* TODO if we dropped a packet then we should introduce a gap here */ - struct frame *const f = frames; - frames = f->next; - free(f); - pthread_cond_broadcast(&cond); +/** @brief Background thread collecting samples + * + * This function collects samples, perhaps converts them to the target format, + * 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; + int n; + struct rtp_header header; + uint16_t seq; + uint32_t timestamp; + struct iovec iov[2]; + + for(;;) { + if(!p) + 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: + continue; + default: + disorder_fatal(errno, "error reading from socket"); + } + } + /* Ignore too-short packets */ + if((size_t)n <= sizeof (struct rtp_header)) { + disorder_info("ignored a short packet"); continue; } - if(samplesOutLeft == 0) { - --nbuffers; - ++ab; - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); + timestamp = htonl(header.timestamp); + seq = htons(header.seq); + /* Ignore packets in the past */ + if(active && lt(timestamp, next_timestamp)) { + disorder_info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, + timestamp, next_timestamp); continue; } - /* Now: (1) there is some data left to read - * (2) there is some space to put it */ - samplesInLeft = frames->nsamples - frames->used; - samplesToCopy = (samplesInLeft < samplesOutLeft - ? samplesInLeft : samplesOutLeft); - memcpy(samplesOut, frame->samples + frames->used, samplesToCopy); - frames->used += samplesToCopy; - samplesOut += samplesToCopy; - samesOutLeft -= samplesToCopy; + /* Ignore packets with the extension bit set. */ + if(header.vpxcc & 0x10) + continue; + p->next = 0; + p->flags = 0; + p->timestamp = timestamp; + /* Convert to target format */ + if(header.mpt & 0x80) + p->flags |= IDLE; + switch(header.mpt & 0x7F) { + case 10: /* L16 */ + p->nsamples = (n - sizeof header) / sizeof(uint16_t); + break; + /* TODO support other RFC3551 media types (when the speaker does) */ + default: + disorder_fatal(0, "unsupported RTP payload type %d", header.mpt & 0x7F); + } + /* See if packet is silent */ + const uint16_t *s = p->samples_raw; + n = p->nsamples; + for(; n > 0; --n) + if(*s++) + break; + if(!n) + p->flags |= SILENT; + 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... */ + if(nsamples >= maxbuffer) { + pthread_mutex_lock(&lock); + while(nsamples >= maxbuffer) { + pthread_cond_wait(&cond, &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; } - pthread_mutex_unlock(&lock); - return 0; } -#endif -void play_rtp(void) { - pthread_t lt; +/** @brief Wait until the buffer is adequately full + * + * Must be called with @ref lock held. + */ +void playrtp_fill_buffer(void) { + /* Discard current buffer contents */ + while(nsamples) { + //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer); + drop_first_packet(); + } + disorder_info("Buffering..."); + /* Wait until there's at least minbuffer samples available */ + while(nsamples < minbuffer) { + //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer); + pthread_cond_wait(&cond, &lock); + } + /* Start from whatever is earliest */ + next_timestamp = pheap_first(&packets)->timestamp; + active = 1; +} - /* We receive and convert audio data in a background thread */ - pthread_create(<, 0, listen_thread, 0); -#if API_ALSA - assert(!"implemented"); -#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 */ - } +/** @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; } -#else -# error No known audio API -#endif + return 0; } /* display usage message and terminate */ static void help(void) { xprintf("Usage:\n" - " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" + " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n" "Options:\n" + " --device, -D DEVICE Output device\n" + " --min, -m FRAMES Buffer low 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" + " --api, -A API Select audio API. Possibilities:\n" + " "); + int first = 1; + for(int n = 0; uaudio_apis[n]; ++n) { + if(uaudio_apis[n]->flags & UAUDIO_API_CLIENT) { + if(first) + first = 0; + else + xprintf(", "); + xprintf("%s", uaudio_apis[n]->name); + } + } + xprintf("\n" + " --command, -e COMMAND Pipe audio to command.\n" + " --pause-mode, -P silence For -e: pauses send silence (default)\n" + " --pause-mode, -P suspend For -e: pauses suspend writes\n" " --help, -h Display usage message\n" " --version, -V Display version number\n" - " --debug, -d Turn on debugging\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); +static size_t playrtp_callback(void *buffer, + size_t max_samples, + void attribute((unused)) *userdata) { + size_t samples; + int silent = 0; + + pthread_mutex_lock(&lock); + /* Get the next packet, junking any that are now in the past */ + const struct packet *p = playrtp_next_packet(); + if(p && contains(p, next_timestamp)) { + /* This packet is ready to play; the desired next timestamp points + * somewhere into it. */ + + /* Timestamp of end of packet */ + const uint32_t packet_end = p->timestamp + p->nsamples; + + /* Offset of desired next timestamp into current packet */ + const uint32_t offset = next_timestamp - p->timestamp; + + /* Pointer to audio data */ + const uint16_t *ptr = (void *)(p->samples_raw + offset); + + /* Compute number of samples left in packet, limited to output buffer + * size */ + samples = packet_end - next_timestamp; + if(samples > max_samples) + samples = max_samples; + + /* Copy into buffer, converting to native endianness */ + size_t i = samples; + int16_t *bufptr = buffer; + while(i > 0) { + *bufptr++ = (int16_t)ntohs(*ptr++); + --i; + } + silent = !!(p->flags & SILENT); + } else { + /* There is no suitable packet. We introduce 0s up to the next packet, or + * to fill the buffer if there's no next packet or that's too many. The + * comparison with max_samples deals with the otherwise troubling overflow + * case. */ + samples = p ? p->timestamp - next_timestamp : max_samples; + if(samples > max_samples) + samples = max_samples; + //info("infill by %zu", samples); + memset(buffer, 0, samples * uaudio_sample_size); + silent = 1; + } + /* Debug dump */ + if(dump_buffer) { + for(size_t i = 0; i < samples; ++i) { + dump_buffer[dump_index++] = ((int16_t *)buffer)[i]; + dump_index %= dump_size; + } + } + /* Advance timestamp */ + next_timestamp += samples; + /* If we're getting behind then try to drop just silent packets + * + * In theory this shouldn't be necessary. The server is supposed to send + * packets at the right rate and compares the number of samples sent with the + * time in order to ensure this. + * + * However, various things could throw this off: + * + * - the server's clock could advance at the wrong rate. This would cause it + * to mis-estimate the right number of samples to have sent and + * inappropriately throttle or speed up. + * + * - playback could happen at the wrong rate. If the playback host's sound + * card has a slightly incorrect clock then eventually it will get out + * of step. + * + * So if we play back slightly slower than the server sends for either of + * these reasons then eventually our buffer, and the socket's buffer, will + * fill, and the kernel will start dropping packets. The result is audible + * and not very nice. + * + * Therefore if we're getting behind, we pre-emptively drop silent packets, + * since a change in the duration of a silence is less noticeable than a + * dropped packet from the middle of continuous music. + * + * (If things go wrong the other way then eventually we run out of packets to + * play and are forced to play silence. This doesn't seem to happen in + * practice but if it does then in the same way we can artificially extend + * silent packets to compensate.) + * + * Dropped packets are always logged; use 'disorder-playrtp --monitor' to + * track how close to target buffer occupancy we are on a once-a-minute + * basis. + */ + if(nsamples > minbuffer && silent) { + disorder_info("dropping %zu samples (%"PRIu32" > %"PRIu32")", + samples, nsamples, minbuffer); + samples = 0; + } + /* Junk obsolete packets */ + playrtp_next_packet(); + pthread_mutex_unlock(&lock); + return samples; +} + +static int compare_family(const struct ifaddrs *a, + const struct ifaddrs *b, + int family) { + int afamily = a->ifa_addr->sa_family; + int bfamily = b->ifa_addr->sa_family; + if(afamily != bfamily) { + /* Preferred family wins */ + if(afamily == family) return 1; + if(bfamily == family) return -1; + /* Either there's no preference or it doesn't help. Prefer IPv4 */ + if(afamily == AF_INET) return 1; + if(bfamily == AF_INET) return -1; + /* Failing that prefer IPv6 */ + if(afamily == AF_INET6) return 1; + if(bfamily == AF_INET6) return -1; + } + return 0; +} + +static int compare_flags(const struct ifaddrs *a, + const struct ifaddrs *b) { + unsigned aflags = a->ifa_flags, bflags = b->ifa_flags; + /* Up interfaces are better than down ones */ + unsigned aup = aflags & IFF_UP, bup = bflags & IFF_UP; + if(aup != bup) + return aup > bup ? 1 : -1; + /* Static addresses are better than dynamic */ + unsigned adynamic = aflags & IFF_DYNAMIC, bdynamic = bflags & IFF_DYNAMIC; + if(adynamic != bdynamic) + return adynamic < bdynamic ? 1 : -1; + unsigned aloopback = aflags & IFF_LOOPBACK, bloopback = bflags & IFF_LOOPBACK; + /* Static addresses are better than dynamic */ + if(aloopback != bloopback) + return aloopback < bloopback ? 1 : -1; + return 0; +} + +static int compare_interfaces(const struct ifaddrs *a, + const struct ifaddrs *b, + int family) { + int c; + if((c = compare_family(a, b, family))) return c; + return compare_flags(a, b); } int main(int argc, char **argv) { - int n; + int n, err; struct addrinfo *res; struct stringlist sl; - const char *sockname; - - static const struct addrinfo prefbind = { - AI_PASSIVE, - PF_INET, - SOCK_DGRAM, - IPPROTO_UDP, - 0, - 0, - 0, - 0 + char *sockname; + int rcvbuf, target_rcvbuf = 0; + socklen_t len; + struct ip_mreq mreq; + struct ipv6_mreq mreq6; + disorder_client *c = NULL; + 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; + pthread_t ltid; + int monitor = 0; + static const int one = 1; + static const struct addrinfo prefs = { + .ai_flags = AI_PASSIVE, + .ai_family = PF_INET, + .ai_socktype = SOCK_DGRAM, + .ai_protocol = IPPROTO_UDP + }; + + /* Timing information is often important to debugging playrtp, so we include + * timestamps in the logs */ + logdate = 1; mem_init(); - if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVd", options, 0)) >= 0) { + if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale"); + while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:MA:", options, 0)) >= 0) { switch(n) { case 'h': help(); - case 'V': version(); + case 'V': version("disorder-playrtp"); case 'd': debugging = 1; break; - default: fatal(0, "invalid option"); + case 'D': uaudio_set("device", optarg); break; + case 'm': minbuffer = 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': + disorder_error(0, "deprecated option; use --api alsa instead"); + backend = &uaudio_alsa; break; +#endif +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + case 'o': + disorder_error(0, "deprecated option; use --api oss instead"); + backend = &uaudio_oss; + break; +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + case 'c': + disorder_error(0, "deprecated option; use --api coreaudio instead"); + backend = &uaudio_coreaudio; + break; +#endif + case 'A': backend = uaudio_find(optarg); break; + case 'C': configfile = optarg; break; + case 's': control_socket = optarg; break; + case 'r': dumpfile = optarg; break; + case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break; + case 'P': uaudio_set("pause-mode", optarg); break; + case 'M': monitor = 1; break; + default: disorder_fatal(0, "invalid option"); } } + if(config_read(0, NULL)) disorder_fatal(0, "cannot read configuration"); + if(!backend) { + backend = uaudio_default(uaudio_apis, UAUDIO_API_CLIENT); + if(!backend) + disorder_fatal(0, "no default uaudio API found"); + disorder_info("default audio API %s", backend->name); + } + if(backend == &uaudio_rtp) { + /* This means that you have NO local sound output. This can happen if you + * use a non-Apple GCC on a Mac (because it doesn't know how to compile + * CoreAudio/AudioHardware.h). */ + disorder_fatal(0, "cannot play RTP through RTP"); + } + if(!maxbuffer) + maxbuffer = 2 * minbuffer; 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 */ - if(!(res = get_address(&sl, &pref, &sockname))) - exit(1); - if((rtpfd = socket(res->ai_family, - res->ai_socktype, - res->ai_protocol)) < 0) - fatal(errno, "error creating socket"); - if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) - fatal(errno, "error binding socket to %s", sockname); - play_rtp(); + 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: + disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]"); + } + disorder_info("version "VERSION" process ID %lu", + (unsigned long)getpid()); + struct sockaddr *addr; + socklen_t addr_len; + if(!strcmp(sl.s[0], "-")) { + /* Pick address family to match known-working connectivity to the server */ + int family = disorder_client_af(c); + /* Get a list of interfaces */ + struct ifaddrs *ifa, *bestifa = NULL; + if(getifaddrs(&ifa) < 0) + disorder_fatal(errno, "error calling getifaddrs"); + /* Try to pick a good one */ + for(; ifa; ifa = ifa->ifa_next) { + if(bestifa == NULL + || compare_interfaces(ifa, bestifa, family) > 0) + bestifa = ifa; + } + if(!bestifa) + disorder_fatal(0, "failed to select a network interface"); + family = bestifa->ifa_addr->sa_family; + if((rtpfd = socket(family, + SOCK_DGRAM, + IPPROTO_UDP)) < 0) + disorder_fatal(errno, "error creating socket (family %d)", family); + /* Bind the address */ + if(bind(rtpfd, bestifa->ifa_addr, + family == AF_INET + ? sizeof (struct sockaddr_in) : sizeof (struct sockaddr_in6)) < 0) + disorder_fatal(errno, "error binding socket"); + static struct sockaddr_storage bound_address; + addr = (struct sockaddr *)&bound_address; + addr_len = sizeof bound_address; + if(getsockname(rtpfd, addr, &addr_len) < 0) + disorder_fatal(errno, "error getting socket address"); + /* Convert to string */ + char addrname[128], portname[32]; + if(getnameinfo(addr, addr_len, + addrname, sizeof addrname, + portname, sizeof portname, + NI_NUMERICHOST|NI_NUMERICSERV) < 0) + disorder_fatal(errno, "getnameinfo"); + /* Ask for audio data */ + if(disorder_rtp_request(c, addrname, portname)) exit(EXIT_FAILURE); + /* Report what we did */ + disorder_info("listening on %s", format_sockaddr(addr)); + } else { + /* Look up address and port */ + if(!(res = get_address(&sl, &prefs, &sockname))) + exit(1); + addr = res->ai_addr; + addr_len = res->ai_addrlen; + /* Create the socket */ + if((rtpfd = socket(res->ai_family, + res->ai_socktype, + res->ai_protocol)) < 0) + disorder_fatal(errno, "error creating socket"); + /* Allow multiple listeners */ + xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one); + is_multicast = multicast(addr); + /* The multicast and unicast/broadcast cases are different enough that they + * are totally split. Trying to find commonality between them causes more + * trouble that it's worth. */ + if(is_multicast) { + /* Stash the multicast group address */ + memcpy(&mgroup, addr, addr_len); + switch(res->ai_addr->sa_family) { + case AF_INET: + mgroup.in.sin_port = 0; + break; + case AF_INET6: + mgroup.in6.sin6_port = 0; + break; + default: + disorder_fatal(0, "unsupported address family %d", + (int)addr->sa_family); + } + /* Bind to to the multicast group address */ + if(bind(rtpfd, addr, addr_len) < 0) + disorder_fatal(errno, "error binding socket to %s", + format_sockaddr(addr)); + /* Add multicast group membership */ + 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) + disorder_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) + disorder_fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); + break; + default: + disorder_fatal(0, "unsupported address family %d", res->ai_family); + } + /* Report what we did */ + disorder_info("listening on %s multicast group %s", + format_sockaddr(addr), format_sockaddr(&mgroup.sa)); + } else { + /* Bind to 0/port */ + switch(addr->sa_family) { + case AF_INET: { + struct sockaddr_in *in = (struct sockaddr_in *)addr; + + memset(&in->sin_addr, 0, sizeof (struct in_addr)); + break; + } + case AF_INET6: { + struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)addr; + + memset(&in6->sin6_addr, 0, sizeof (struct in6_addr)); + break; + } + default: + disorder_fatal(0, "unsupported family %d", (int)addr->sa_family); + } + if(bind(rtpfd, addr, addr_len) < 0) + disorder_fatal(errno, "error binding socket to %s", + format_sockaddr(addr)); + /* Report what we did */ + disorder_info("listening on %s", format_sockaddr(addr)); + } + } + len = sizeof rcvbuf; + if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) + disorder_fatal(errno, "error calling getsockopt SO_RCVBUF"); + if(target_rcvbuf > rcvbuf) { + if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, + &target_rcvbuf, sizeof target_rcvbuf) < 0) + disorder_error(errno, "error calling setsockopt SO_RCVBUF %d", + target_rcvbuf); + /* We try to carry on anyway */ + else + disorder_info("changed socket receive buffer from %d to %d", + rcvbuf, target_rcvbuf); + } else + disorder_info("default socket receive buffer %d", rcvbuf); + //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer); + if(logfp) + disorder_info("WARNING: -L option can impact performance"); + if(control_socket) { + pthread_t tid; + + if((err = pthread_create(&tid, 0, control_thread, 0))) + disorder_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) + disorder_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) + disorder_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) + disorder_fatal(errno, "mapping %s", dumpfile); + disorder_info("dumping to %s", dumpfile); + } + /* Set up output. Currently we only support L16 so there's no harm setting + * the format before we know what it is! */ + uaudio_set_format(44100/*Hz*/, 2/*channels*/, + 16/*bits/channel*/, 1/*signed*/); + uaudio_set("application", "disorder-playrtp"); + backend->start(playrtp_callback, NULL); + /* We receive and convert audio data in a background thread */ + if((err = pthread_create(<id, 0, listen_thread, 0))) + disorder_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))) + disorder_fatal(err, "pthread_create queue_thread"); + pthread_mutex_lock(&lock); + time_t lastlog = 0; + for(;;) { + /* Wait for the buffer to fill up a bit */ + playrtp_fill_buffer(); + /* Start playing now */ + disorder_info("Playing..."); + next_timestamp = pheap_first(&packets)->timestamp; + active = 1; + pthread_mutex_unlock(&lock); + backend->activate(); + pthread_mutex_lock(&lock); + /* Wait until the buffer empties out + * + * If there's a packet that we can play right now then we definitely + * continue. + * + * Also if there's at least minbuffer samples we carry on regardless and + * insert silence. The assumption is there's been a pause but more data + * is now available. + */ + while(nsamples >= minbuffer + || (nsamples > 0 + && contains(pheap_first(&packets), next_timestamp))) { + if(monitor) { + time_t now = xtime(0); + + if(now >= lastlog + 60) { + int offset = nsamples - minbuffer; + double offtime = (double)offset / (uaudio_rate * uaudio_channels); + disorder_info("%+d samples off (%d.%02ds, %d bytes)", + offset, + (int)fabs(offtime) * (offtime < 0 ? -1 : 1), + (int)(fabs(offtime) * 100) % 100, + offset * uaudio_bits / CHAR_BIT); + lastlog = now; + } + } + //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer); + pthread_cond_wait(&cond, &lock); + } +#if 0 + if(nsamples) { + struct packet *p = pheap_first(&packets); + fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n", + nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples); + } +#endif + /* Stop playing for a bit until the buffer re-fills */ + pthread_mutex_unlock(&lock); + backend->deactivate(); + pthread_mutex_lock(&lock); + active = 0; + /* Go back round */ + } return 0; }