X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/9ae1516d1575f0ed0dc6c599ccbf81c1f3661253..ceb044f4e944cd32110619d50b8ccb6dfddd3ac1:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index b5835c4..fe9d1e2 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -51,6 +51,9 @@ /** @brief RTP socket */ static int rtpfd; +/** @brief Log output */ +static FILE *logfp; + /** @brief Output device */ static const char *device; @@ -60,7 +63,7 @@ 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 */ @@ -70,16 +73,19 @@ 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 = 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 */ #define INFILL_SAMPLES (44100 * 2) /* 1s */ -#define MAXBUFFER (3 * 88200) /* maximum buffer contents */ - /** @brief Received packet * * Packets are recorded in an ordered linked list. */ @@ -108,7 +114,8 @@ static unsigned long nsamples; /** @brief Linked list of packets * - * In ascending order of timestamp. */ + * In ascending order of timestamp. Really this should be a heap for more + * efficient access. */ static struct packet *packets; /** @brief Timestamp of next packet to play. @@ -135,6 +142,7 @@ 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 } }; @@ -159,6 +167,15 @@ 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_first_packet(void) { + struct packet *const p = packets; + packets = p->next; + nsamples -= p->nsamples; + free(p); + pthread_cond_broadcast(&cond); +} + /** @brief Background thread collecting samples * * This function collects samples, perhaps converts them to the target format, @@ -186,8 +203,10 @@ 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->timestamp = ntohl(packet.header.timestamp); /* Ignore packets in the past */ if(active && lt(p->timestamp, next_timestamp)) { @@ -201,8 +220,12 @@ static void *listen_thread(void attribute((unused)) *arg) { 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); + { + size_t i; + + for(i = 0; i < p->nsamples; ++i) + p->samples_float[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767); + } #else /* ALSA can do any necessary conversion itself (though it might be better * to do any necessary conversion in the background) */ @@ -214,13 +237,20 @@ static void *listen_thread(void attribute((unused)) *arg) { fatal(0, "unsupported RTP payload type %d", packet.header.mpt & 0x7F); } + if(logfp) + fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", + ntohs(packet.header.seq), + p->timestamp, p->nsamples, p->timestamp + p->nsamples); pthread_mutex_lock(&lock); /* 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); + if(nsamples >= maxbuffer) { + info("buffer full"); + while(nsamples >= maxbuffer) + pthread_cond_wait(&cond, &lock); + } for(pp = &packets; *pp && lt((*pp)->timestamp, p->timestamp); pp = &(*pp)->next) @@ -245,48 +275,76 @@ static void *listen_thread(void attribute((unused)) *arg) { #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; 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) { + if(packets) { + /* There's a packet */ + const uint32_t packet_start = packets->timestamp; + const uint32_t packet_end = packets->timestamp + packets->nsamples; + + if(le(packet_end, next_timestamp)) { + /* This packet is in the past */ + info("dropping buffered past packet %"PRIx32" < %"PRIx32, + packet_start, next_timestamp); + drop_first_packet(); + continue; + } + if(ge(next_timestamp, packet_start) + && lt(next_timestamp, packet_end)) { + /* This packet is suitable */ + const uint32_t offset = next_timestamp - packet_start; + uint32_t samples_available = packet_end - next_timestamp; + if(samples_available > samplesOutLeft) + samples_available = samplesOutLeft; + memcpy(samplesOut, + packets->samples_float + offset, + samples_available * sizeof(float)); + samplesOut += samples_available; + next_timestamp += samples_available; + samplesOutLeft -= samples_available; + if(ge(next_timestamp, packet_end)) + drop_first_packet(); + continue; + } + } + /* We didn't find a suitable packet (though there might still be + * unsuitable ones). We infill with 0s. */ + if(packets) { + /* There is a next packet, only infill up to that point */ + uint32_t samples_available = packets->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; @@ -404,17 +462,9 @@ static void play_rtp(void) { } if(packets && ge(next_timestamp, packets->timestamp + packets->nsamples)) { - struct packet *p = packets; - info("dropping buffered past packet %"PRIx32" < %"PRIx32, packets->timestamp, next_timestamp); - - packets = p->next; - if(packets) - assert(lt(p->timestamp, packets->timestamp)); - nsamples -= p->nsamples; - free(p); - pthread_cond_broadcast(&cond); + drop_first_packet(); continue; } /* Wait for ALSA to ask us for more data */ @@ -460,17 +510,8 @@ static void play_rtp(void) { } else { samples_written = frames_written * 2; next_timestamp += samples_written; - if(ge(next_timestamp, packet_end)) { - /* We're done with this packet */ - struct packet *p = packets; - - packets = p->next; - if(packets) - assert(lt(p->timestamp, packets->timestamp)); - nsamples -= p->nsamples; - free(p); - pthread_cond_broadcast(&cond); - } + if(ge(next_timestamp, packet_end)) + drop_first_packet(); infilling = 0; } } else { @@ -562,14 +603,14 @@ 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); @@ -578,9 +619,13 @@ 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->timestamp; + active = 1; status = AudioDeviceStart(adid, adioproc); if(status) fatal(0, "AudioDeviceStart: %d", (int)status); @@ -591,6 +636,7 @@ static void play_rtp(void) { status = AudioDeviceStop(adid, adioproc); if(status) fatal(0, "AudioDeviceStop: %d", (int)status); + active = 0; /* Go back round */ } } @@ -604,12 +650,13 @@ 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); } @@ -640,7 +687,7 @@ 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(); @@ -648,9 +695,13 @@ 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)