X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/e83d0967d4c0965eb8036248acc20d1bf12ad1d8..28bacdc0e8ab3e4d8b7f628f59d65e4fa38b9622:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index 7ec35d0..86d8337 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -17,6 +17,11 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 * USA */ +/** @file clients/playrtp.c + * @brief RTP player + * + * This RTP player supports Linux (ALSA) and Darwin (Core Audio) systems. + */ #include #include "types.h" @@ -29,6 +34,9 @@ #include #include #include +#include +#include +#include #include "log.h" #include "mem.h" @@ -36,68 +44,252 @@ #include "addr.h" #include "syscalls.h" #include "rtp.h" -#include "debug.h" +#include "defs.h" +#include "vector.h" +#include "heap.h" #if HAVE_COREAUDIO_AUDIOHARDWARE_H # include #endif +#if API_ALSA +#include +#endif +#define readahead linux_headers_are_borked + +/** @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 */ +static const char *device; + +/** @brief Maximum samples per packet we'll support + * + * NB that two channels = two samples in this program. + */ +#define MAXSAMPLES 2048 + +/** @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 */ + +/** @brief Maximum sample size + * + * The maximum supported size (in bytes) of one sample. */ +#define MAXSAMPLESIZE 2 + +/** @brief Buffer high watermark + * + * We'll only start playing when this many samples are available. */ +static unsigned readahead = 2 * 2 * 44100; + +/** @brief Maximum buffer size + * + * 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 + * + * This is an upper bound - in practice we expxect the underlying audio API to + * only ask for a much smaller number of samples in any one go. + */ +#define INFILL_SAMPLES (44100 * 2) /* 1s */ + +/** @brief Received packet + * + * Received packets are kept in a binary heap (see @ref pheap) ordered by + * timestamp. + */ +struct packet { + /** @brief Number of samples in this packet */ + uint32_t nsamples; + /** @brief Timestamp from RTP packet + * + * NB that "timestamps" are really sample counters. Use lt() or lt_packet() + * to compare timestamps. + */ + uint32_t timestamp; + /** @brief Raw sample data + * + * Only the first @p nsamples samples are defined; the rest is uninitialized + * data. + */ + unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE]; }; -static unsigned long nsamples; /* total samples available */ +/** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic + * + * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$. + * + * See also lt_packet(). + */ +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 Ordering for packets, used by @ref pheap */ +static inline int lt_packet(const struct packet *a, const struct packet *b) { + return lt(a->timestamp, b->timestamp); +} + +/** @struct pheap + * @brief Binary heap of packets ordered by timestamp */ +HEAP_TYPE(pheap, struct packet *, lt_packet); + +/** @brief Binary heap of received packets */ +static struct pheap packets; + +/** @brief Total number of samples available */ +static unsigned long 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; + +/** @brief True if actively playing + * + * This is true when playing and false when just buffering. */ +static int active; + +/** @brief Structure of free packet list */ +union free_packet { + struct packet p; + union free_packet *next; +}; + +/** @brief Linked list of free packets + * + * This is a linked list of formerly used packets. For preference we re-use + * packets that have already been used rather than unused ones, to limit the + * size of the program's working set. If there are no free packets in the list + * we try @ref next_free_packet instead. + * + * Must hold @ref lock when accessing this. + */ +static union free_packet *free_packets; + +/** @brief Array of new free packets + * + * There are @ref count_free_packets ready to use at this address. If there + * are none left we allocate more memory. + * + * Must hold @ref lock when accessing this. + */ +static union free_packet *next_free_packet; + +/** @brief Count of new free packets at @ref next_free_packet + * + * Must hold @ref lock when accessing this. + */ +static size_t count_free_packets; -static struct frame *frames; /* received frames in ascending order - * of timestamp */ +/** @brief Lock protecting @ref packets + * + * This also protects the packet memory allocation infrastructure, @ref + * free_packets and @ref next_free_packet. */ static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; -/* lock protecting frame list */ -static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER; -/* signalled whenever we add a new frame */ +/** @brief Condition variable signalled whenever @ref packets is changed */ +static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; 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' }, { 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; +/** @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; } -/* 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 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 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; + free_packet(p); + 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. */ +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(!f) - f = xmalloc(sizeof *f); - 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: @@ -106,93 +298,360 @@ static listen_thread(void attribute((unused)) *arg) { fatal(errno, "error reading from socket"); } } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H + /* Ignore too-short packets */ + if((size_t)n <= sizeof (struct rtp_header)) { + info("ignored a short packet"); + continue; + } + timestamp = htonl(header.timestamp); + seq = htons(header.seq); + /* Ignore packets in the past */ + 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.mtp & 0x7F) { + switch(header.mpt & 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); + 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) */ + /* 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); + fatal(0, "unsupported RTP payload type %d", + header.mpt & 0x7F); } -#endif - f->used = 0; - f->timestamp = ntohl(packet.header.timestamp); - 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; + 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) { + info("buffer full"); + while(nsamples >= maxbuffer) + pthread_cond_wait(&cond, &lock); + } + /* Add the packet to the heap */ + pheap_insert(&packets, p); + nsamples += p->nsamples; pthread_cond_broadcast(&cond); pthread_mutex_unlock(&lock); - f = 0; } } +/** @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 -static OSStatus adioproc(AudioDeviceID inDevice, - const AudioTimeStamp *inNow, - const AudioBufferList *inInputData, - const AudioTimeStamp *inInputTime, - AudioBufferList *outOutputData, - const AudioTimeStamp *inOutputTime, - void *inClientData) { +/** @brief Callback from Core Audio */ +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; - - 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); - continue; - } - if(samplesOutLeft == 0) { - --nbuffers; - ++ab; - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); - continue; + const struct packet *p; + uint32_t samples_available; + + pthread_mutex_lock(&lock); + 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(pheap_count(&packets)) { + 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.) */ + break; + } + p = pheap_count(&packets) ? pheap_first(&packets) : 0; + if(p && contains(p, next_timestamp)) { + /* This packet is ready to play */ + 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)); + + 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 - that'll be dealt with next time + * round */ + } else { + /* No packet is ready to play (and there might be no packet at all) */ + samples_available = p ? p->timestamp - next_timestamp + : samplesOutLeft; + if(samples_available > samplesOutLeft) + samples_available = samplesOutLeft; + info("infill by %"PRIu32, samples_available); + /* Conveniently the buffer is 0 to start with */ + next_timestamp += samples_available; + samplesOut += samples_available; + samplesOutLeft -= samples_available; + } } - /* 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; + ++ab; + --nbuffers; } pthread_mutex_unlock(&lock); return 0; } #endif -void play_rtp(void) { - pthread_t lt; +/** @brief Play an RTP stream + * + * This is the guts of the program. It is responsible for: + * - starting the listening thread + * - opening the audio device + * - reading ahead to build up a buffer + * - arranging for audio to be played + * - detecting when the buffer has got too small and re-buffering + */ +static void play_rtp(void) { + pthread_t ltid; /* We receive and convert audio data in a background thread */ - pthread_create(<, 0, listen_thread, 0); + pthread_create(<id, 0, listen_thread, 0); #if API_ALSA - assert(!"implemented"); + { + 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; + int infilling = 0, escape = 0; + time_t logged, now; + uint32_t packet_start, packet_end; + + /* 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 */ + + 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); + if(!prepared) { + if((err = snd_pcm_prepare(pcm))) + fatal(0, "error calling snd_pcm_prepare: %d", err); + prepared = 1; + } + 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 && !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); + 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 */ + 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 + offset, + frames_available); + if(frames_written < 0) { + 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; + next_timestamp += samples_written; + 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[INFILL_SAMPLES]; + size_t samples_available = INFILL_SAMPLES, frames_available; + + /* 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) { + 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; + 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_nonblock(pcm, 0))) + fatal(0, "error calling snd_pcm_nonblock: %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; + pthread_mutex_lock(&lock); + } + + } #elif HAVE_COREAUDIO_AUDIOHARDWARE_H { OSStatus status; @@ -218,14 +677,14 @@ 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); @@ -234,19 +693,24 @@ void play_rtp(void) { pthread_mutex_lock(&lock); for(;;) { /* Wait for the buffer to fill up a bit */ - while(nsamples < READAHEAD) + info("Buffering..."); + while(nsamples < readahead) pthread_cond_wait(&cond, &lock); /* Start playing now */ + info("Playing..."); + next_timestamp = pheap_first(&packets)->timestamp; + active = 1; status = AudioDeviceStart(adid, adioproc); if(status) fatal(0, "AudioDeviceStart: %d", (int)status); /* Wait until the buffer empties out */ - while(nsamples >= MINBUFFER) + 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); + active = 0; /* Go back round */ } } @@ -260,9 +724,13 @@ 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" " --help, -h Display usage message\n" " --version, -V Display version number\n" - " --debug, -d Turn on debugging\n"); + ); xfclose(stdout); exit(0); } @@ -278,9 +746,9 @@ int main(int argc, char **argv) { int n; struct addrinfo *res; struct stringlist sl; - const char *sockname; + char *sockname; - static const struct addrinfo prefbind = { + static const struct addrinfo prefs = { AI_PASSIVE, PF_INET, SOCK_DGRAM, @@ -293,14 +761,21 @@ int main(int argc, char **argv) { mem_init(); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVd", 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(); 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; default: fatal(0, "invalid option"); } } + if(!maxbuffer) + maxbuffer = 4 * readahead; argc -= optind; argv += optind; if(argc < 1 || argc > 2) @@ -308,7 +783,7 @@ int main(int argc, char **argv) { sl.n = argc; sl.s = argv; /* Listen for inbound audio data */ - if(!(res = get_address(&sl, &pref, &sockname))) + if(!(res = get_address(&sl, &prefs, &sockname))) exit(1); if((rtpfd = socket(res->ai_family, res->ai_socktype,