X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/1153fd2325f924dcd8990b41ca67a4d683394e48..ff8ddcef6032bcd20fa2b8097ab9ad7d785cc3c8:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index e259b2c..398f47b 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "log.h" #include "mem.h" @@ -51,6 +52,9 @@ /** @brief RTP socket */ static int rtpfd; +/** @brief Log output */ +static FILE *logfp; + /** @brief Output device */ static const char *device; @@ -60,7 +64,7 @@ static const char *device; */ #define MAXSAMPLES 2048 -/** @brief Minimum buffer size +/** @brief Minimum low watermark * * We'll stop playing if there's only this many samples in the buffer. */ static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ @@ -70,45 +74,42 @@ static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ * The maximum supported size (in bytes) of one sample. */ #define MAXSAMPLESIZE 2 -/** @brief Buffer size +/** @brief Buffer high watermark * * We'll only start playing when this many samples are available. */ -static unsigned readahead = 4 * 2 * 44100; /* 4 seconds */ - -#define MAXBUFFER (3 * 88200) /* maximum buffer contents */ +static unsigned readahead = 2 * 2 * 44100; -/** @brief Received packet +/** @brief Maximum buffer size * - * Packets are recorded in an ordered linked list. */ + * We'll stop reading from the network if we have this many samples. */ +static unsigned maxbuffer; + +/** @brief Number of samples to infill by in one go */ +#define INFILL_SAMPLES (44100 * 2) /* 1s */ + +/** @brief Received packet */ 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; + uint32_t nsamples; /** @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 }; /** @brief Total number of samples available */ static unsigned long nsamples; -/** @brief Linked list of packets +/** @brief Mapping of sequence numbers to packets * - * In ascending order of timestamp. */ -static struct packet *packets; + * This isn't very efficient - 256KB on 32-bit machines, 512KB if you do a + * 64-bit build for some reason. It can be optimized later if need be. */ +static struct packet *packets[65536]; + +/** @brief Total number of packets */ +static unsigned npackets; /** @brief Timestamp of next packet to play. * @@ -122,6 +123,24 @@ static uint32_t next_timestamp; * This is true when playing and false when just buffering. */ static int active; +/** @brief Sequence number of next packet we expxect to play */ +static uint16_t sequence; + +/** @brief Structure of free packet list */ +union free_packet { + struct packet p; + union free_packet *next; +}; + +/** @brief Linked list of free packets */ +static union free_packet *free_packets; + +/** @brief Array of new free packets */ +static union free_packet *next_free_packet; + +/** @brief Count of new free packets */ +static size_t count_free_packets; + /** @brief Lock protecting @ref packets */ static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; @@ -134,33 +153,94 @@ static const struct option options[] = { { "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' }, { 0, 0, 0, 0 } }; +/** @brief Return a new packet + * + * Assumes that @ref lock is held. */ +static struct packet *new_packet(void) { + struct packet *p; + + if(free_packets) { + p = &free_packets->p; + free_packets = free_packets->next; + } else { + if(!count_free_packets) { + next_free_packet = xcalloc(1024, sizeof (union free_packet)); + count_free_packets = 1024; + } + p = &(next_free_packet++)->p; + --count_free_packets; + } + return p; +} + +/** @brief Free a packet + * + * Assumes that @ref lock is held. */ +static void free_packet(struct packet *p) { + union free_packet *u = (union free_packet *)p; + u->next = free_packets; + free_packets = u; +} + /** @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 Return true iff a >= b in sequence-space arithmetic */ +static inline int ge(uint32_t a, uint32_t b) { + return !lt(a, b); +} + +/** @brief Return true iff a > b in sequence-space arithmetic */ +static inline int gt(uint32_t a, uint32_t b) { + return lt(b, a); +} + +/** @brief Return true iff a <= b in sequence-space arithmetic */ +static inline int le(uint32_t a, uint32_t b) { + return !lt(b, a); +} + +/** @brief Drop the packet at the head of the queue */ +static void drop_packet(unsigned sequence) { + if(packets[sequence]) { + nsamples -= packets[sequence]->nsamples; + free_packet(packets[sequence]); + packets[sequence] = 0; + pthread_cond_broadcast(&cond); + --npackets; + } +} + /** @brief Background thread collecting samples * * This function collects samples, perhaps converts them to the target format, * and adds them to the packet list. */ 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); + if(!p) { + pthread_mutex_lock(&lock); + p = new_packet(); + pthread_mutex_unlock(&lock); + } + 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; + n = readv(rtpfd, iov, 2); if(n < 0) { switch(errno) { case EINTR: @@ -170,102 +250,141 @@ 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; + } + pthread_mutex_lock(&lock); + p = new_packet(); + p->timestamp = timestamp; /* Convert to target format */ - switch(packet.header.mpt & 0x7F) { + 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 + p->nsamples = (n - sizeof header) / sizeof(uint16_t); /* 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 + /* TODO we could readv into the buffer */ 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) { + info("buffer full"); + while(nsamples >= maxbuffer) + pthread_cond_wait(&cond, &lock); } + /* If there's a packet there already we overwrite it; perhaps it is left + * over from an earlier stage. */ + drop_packet(seq); + /* Record this packet */ + packets[seq] = p; + /* If we currently have no idea where to start playing, this is it */ + if(!npackets) + sequence = seq; + ++npackets; + nsamples += p->nsamples; + pthread_cond_broadcast(&cond); pthread_mutex_unlock(&lock); } } +/** @brief Return true if @p p contains @p timestamp */ +static inline int contains(const struct packet *p, uint32_t timestamp) { + const uint32_t packet_start = p->timestamp; + const uint32_t packet_end = p->timestamp + p->nsamples; + + return (ge(timestamp, packet_start) + && lt(timestamp, packet_end)); +} + #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) { +static OSStatus adioproc + (AudioDeviceID attribute((unused)) inDevice, + const AudioTimeStamp attribute((unused)) *inNow, + const AudioBufferList attribute((unused)) *inInputData, + const AudioTimeStamp attribute((unused)) *inInputTime, + AudioBufferList *outOutputData, + const AudioTimeStamp attribute((unused)) *inOutputTime, + void attribute((unused)) *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; + const struct packet *p; 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; + while(nbuffers > 0) { + float *samplesOut = ab->mData; + size_t samplesOutLeft = ab->mDataByteSize / sizeof (float); + + while(samplesOutLeft > 0) { + /* Look for a suitable packet, dropping any unsuitable ones along the + * way. Unsuitable packets are ones that are in the past. */ + while(npackets + && (!packets[sequence] + || le(packets[sequence]->timestamp + + packets[sequence]->nsamples, + next_timestamp))) + drop_packet(sequence++); + p = packets[sequence]; + if(p) { + if(contains(p, next_timestamp)) { + /* This packet is suitable */ + const uint32_t packet_end = p->timestamp + p->nsamples; + const uint32_t offset = next_timestamp - p->timestamp; + const uint16_t *ptr = + (void *)(p->samples_raw + offset * sizeof (uint16_t)); + uint32_t samples_available = packet_end - next_timestamp; + if(samples_available > samplesOutLeft) + samples_available = samplesOutLeft; + next_timestamp += samples_available; + samplesOutLeft -= samples_available; + while(samples_available-- > 0) + *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767); + /* We don't bother junking the packet or advancing sequence - that'll + * be dealt with next time round */ + continue; + } + } + /* We didn't find a suitable packet (though there might still be + * unsuitable ones). We infill with 0s. */ + if(p) { + /* There is a next packet, only infill up to that point */ + uint32_t samples_available = p->timestamp - next_timestamp; + + if(samples_available > samplesOutLeft) + samples_available = samplesOutLeft; + info("infill by %"PRIu32, samples_available); + /* Convniently the buffer is 0 to start with */ + next_timestamp += samples_available; + samplesOut += samples_available; + samplesOutLeft -= samples_available; + } else { + /* There's no next packet at all */ + info("infilled by %zu", samplesOutLeft); + next_timestamp += samplesOutLeft; + samplesOut += samplesOutLeft; + samplesOutLeft = 0; + } } - /* 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; + ++ab; + --nbuffers; } pthread_mutex_unlock(&lock); return 0; @@ -303,7 +422,9 @@ static void play_rtp(void) { size_t samples_written; int prepared = 1; int err; - int infilling = 0; + int infilling = 0, escape = 0; + time_t logged, now; + uint32_t packet_start, packet_end; /* Open ALSA */ if((err = snd_pcm_open(&pcm, @@ -347,9 +468,13 @@ static void play_rtp(void) { /* Ready to go */ + time(&logged); pthread_mutex_lock(&lock); for(;;) { /* Wait for the buffer to fill up a bit */ + logged = now; + info("%lu samples in buffer (%lus)", nsamples, + nsamples / (44100 * 2)); info("Buffering..."); while(nsamples < readahead) pthread_cond_wait(&cond, &lock); @@ -358,66 +483,121 @@ static void play_rtp(void) { fatal(0, "error calling snd_pcm_prepare: %d", err); prepared = 1; } + assert(sequence != -1); /* Start at the first available packet */ - next_timestamp = packets->timestamp; + next_timestamp = packets[sequence]->timestamp; active = 1; infilling = 0; + escape = 0; + logged = now; + info("%lu samples in buffer (%lus)", nsamples, + nsamples / (44100 * 2)); info("Playing..."); /* Wait until the buffer empties out */ - while(nsamples >= minbuffer) { + while(nsamples >= minbuffer && !escape) { + time(&now); + if(now > logged + 10) { + logged = now; + info("%lu samples in buffer (%lus)", nsamples, + nsamples / (44100 * 2)); + } + if(packets + && ge(next_timestamp, packets->timestamp + packets->nsamples)) { + info("dropping buffered past packet %"PRIx32" < %"PRIx32, + packets->timestamp, next_timestamp); + drop_first_packet(); + continue; + } /* Wait for ALSA to ask us for more data */ pthread_mutex_unlock(&lock); - snd_pcm_wait(pcm, -1); + write(2, ".", 1); /* TODO remove me sometime */ + switch(err = snd_pcm_wait(pcm, -1)) { + case 0: + info("snd_pcm_wait timed out"); + break; + case 1: + break; + default: + fatal(0, "snd_pcm_wait returned %d", err); + } 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; + packet_start = packets->timestamp; + packet_end = packets->timestamp + packets->nsamples; + if(ge(next_timestamp, packet_start) + && lt(next_timestamp, packet_end)) { + /* The target timestamp is somewhere in this packet */ + const uint32_t offset = next_timestamp - packets->timestamp; + const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp; const size_t frames_available = samples_available / 2; frames_written = snd_pcm_writei(pcm, - packets->samples_raw + packets->nused, + packets->samples_raw + offset, frames_available); if(frames_written < 0) { - if(frames_written != -EAGAIN) + switch(frames_written) { + case -EAGAIN: + info("snd_pcm_wait() returned but we got -EAGAIN!"); + break; + case -EPIPE: + error(0, "error calling snd_pcm_writei: %ld", + (long)frames_written); + escape = 1; + break; + default: fatal(0, "error calling snd_pcm_writei: %ld", (long)frames_written); + } } else { 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); - } + if(ge(next_timestamp, packet_end)) + drop_first_packet(); infilling = 0; } } 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; + static const uint16_t zeros[INFILL_SAMPLES]; + size_t samples_available = INFILL_SAMPLES, frames_available; - if(!infilling) { - info("Infilling..."); - infilling = 1; - } - if(packets && next_timestamp + samples_available > packets->timestamp) + /* If the maximum infill would take us past the start of the next + * packet then we truncate the infill to the right amount. */ + if(lt(packets->timestamp, + next_timestamp + samples_available)) samples_available = packets->timestamp - next_timestamp; + if((int)samples_available < 0) { + info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32, + packets->timestamp, next_timestamp, + next_timestamp + INFILL_SAMPLES, samples_available); + } frames_available = samples_available / 2; + if(!infilling) { + info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]", + samples_available, next_timestamp, + packets->timestamp, packets->timestamp + packets->nsamples); + //infilling++; + } frames_written = snd_pcm_writei(pcm, zeros, frames_available); if(frames_written < 0) { - if(frames_written != -EAGAIN) + switch(frames_written) { + case -EAGAIN: + info("snd_pcm_wait() returned but we got -EAGAIN!"); + break; + case -EPIPE: + error(0, "error calling snd_pcm_writei: %ld", + (long)frames_written); + escape = 1; + break; + default: fatal(0, "error calling snd_pcm_writei: %ld", (long)frames_written); - } else + } + } else { + samples_written = frames_written * 2; next_timestamp += samples_written; + } } } active = 0; @@ -425,8 +605,13 @@ static void play_rtp(void) { pthread_mutex_unlock(&lock); if((err = snd_pcm_nonblock(pcm, 0))) fatal(0, "error calling snd_pcm_nonblock: %d", err); - if((err = snd_pcm_drain(pcm))) - fatal(0, "error calling snd_pcm_drain: %d", err); + if(escape) { + if((err = snd_pcm_drop(pcm))) + fatal(0, "error calling snd_pcm_drop: %d", err); + escape = 0; + } else + if((err = snd_pcm_drain(pcm))) + fatal(0, "error calling snd_pcm_drain: %d", err); if((err = snd_pcm_nonblock(pcm, 1))) fatal(0, "error calling snd_pcm_nonblock: %d", err); prepared = 0; @@ -459,14 +644,14 @@ static void play_rtp(void) { 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)); + D(("mFormatID %08lx", asbd.mFormatID)); + D(("mFormatFlags %08lx", asbd.mFormatFlags)); + D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket)); + D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket)); + D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame)); + D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame)); + D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel)); + D(("mReserved %08lx", asbd.mReserved)); if(asbd.mFormatID != kAudioFormatLinearPCM) fatal(0, "audio device does not support kAudioFormatLinearPCM"); status = AudioDeviceAddIOProc(adid, adioproc, 0); @@ -475,9 +660,13 @@ static void play_rtp(void) { pthread_mutex_lock(&lock); for(;;) { /* Wait for the buffer to fill up a bit */ + info("Buffering..."); while(nsamples < readahead) pthread_cond_wait(&cond, &lock); /* Start playing now */ + info("Playing..."); + next_timestamp = packets[sequence]->timestamp; + active = 1; status = AudioDeviceStart(adid, adioproc); if(status) fatal(0, "AudioDeviceStart: %d", (int)status); @@ -488,6 +677,7 @@ static void play_rtp(void) { status = AudioDeviceStop(adid, adioproc); if(status) fatal(0, "AudioDeviceStop: %d", (int)status); + active = 0; /* Go back round */ } } @@ -501,12 +691,13 @@ static void help(void) { xprintf("Usage:\n" " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" "Options:\n" - " --help, -h Display usage message\n" - " --version, -V Display version number\n" - " --debug, -d Turn on debugging\n" " --device, -D DEVICE Output device\n" " --min, -m FRAMES Buffer low water mark\n" - " --buffer, -b FRAMES Buffer high water mark\n"); + " --buffer, -b FRAMES Buffer high water mark\n" + " --max, -x FRAMES Buffer maximum size\n" + " --help, -h Display usage message\n" + " --version, -V Display version number\n" + ); xfclose(stdout); exit(0); } @@ -537,7 +728,7 @@ int main(int argc, char **argv) { mem_init(); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVdD:m:b:", options, 0)) >= 0) { + while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) { switch(n) { case 'h': help(); case 'V': version(); @@ -545,9 +736,13 @@ int main(int argc, char **argv) { 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; default: fatal(0, "invalid option"); } } + if(!maxbuffer) + maxbuffer = 4 * readahead; argc -= optind; argv += optind; if(argc < 1 || argc > 2)