/** @file clients/playrtp.c
* @brief RTP player
*
- * This RTP player supports Linux (ALSA) and Darwin (Core Audio) systems.
+ * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
+ * and Apple Mac (<a
+ * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
+ * systems. There is no support for Microsoft Windows yet, and that will in
+ * fact probably an entirely separate program.
+ *
+ * The program runs (at least) two threads. listen_thread() is responsible for
+ * reading RTP packets off the wire and adding them to the binary heap @ref
+ * packets, assuming they are basically sound.
+ *
+ * The main thread is responsible for actually playing audio. In ALSA this
+ * means it waits until ALSA says it's ready for more audio which it then
+ * plays.
+ *
+ * InCore Audio the main thread is only responsible for starting and stopping
+ * play: the system does the actual playback in its own private thread, and
+ * calls adioproc() to fetch the audio data.
+ *
+ * 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.
*/
#include <config.h>
* 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. */
/** @brief Flags
*
* Valid values are:
- * - @ref IDLE: the idle bit was set in the RTP packet
+ * - @ref IDLE - the idle bit was set in the RTP packet
*/
unsigned flags;
-#define IDLE 0x0001 /**< idle bit set in RTP packet */
+/** @brief idle bit set in RTP packet*/
+#define IDLE 0x0001
/** @brief Raw sample data
*
* Only the first @p nsamples samples are defined; the rest is uninitialized
* data.
*/
- unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
+ uint16_t samples_raw[MAXSAMPLES];
};
/** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic
/** @brief Background thread collecting samples
*
* This function collects samples, perhaps converts them to the target format,
- * and adds them to the packet list. */
+ * and adds them to the packet list.
+ *
+ * It is crucial that the gap between successive calls to read() is as small as
+ * possible: otherwise packets will be dropped.
+ *
+ * We use a binary heap to ensure that the unavoidable effort is at worst
+ * logarithmic in the total number of packets - in fact if packets are mostly
+ * received in order then we will largely do constant work per packet since the
+ * newest packet will always be last.
+ *
+ * Of more concern is that we must acquire the lock on the heap to add a packet
+ * to it. If this proves a problem in practice then the answer would be
+ * (probably doubly) linked list with new packets added the end and a second
+ * thread which reads packets off the list and adds them to the heap.
+ *
+ * We keep memory allocation (mostly) very fast by keeping pre-allocated
+ * packets around; see @ref new_packet().
+ */
static void *listen_thread(void attribute((unused)) *arg) {
struct packet *p = 0;
int n;
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;
+ iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
n = readv(rtpfd, iov, 2);
if(n < 0) {
switch(errno) {
switch(header.mpt & 0x7F) {
case 10:
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:
* 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");
+ //info("Buffer full");
+ write(2, "B", 1);
while(nsamples >= maxbuffer)
pthread_cond_wait(&cond, &lock);
}
}
}
-/** @brief Return true if @p p contains @p timestamp */
+/** @brief Return true if @p p contains @p timestamp
+ *
+ * Containment implies that a sample @p timestamp exists within the packet.
+ */
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;
&& lt(timestamp, packet_end));
}
+/** @brief Wait until the buffer is adequately full
+ *
+ * Must be called with @ref lock held.
+ */
+static void fill_buffer(void) {
+ info("Buffering...");
+ while(nsamples < readahead)
+ pthread_cond_wait(&cond, &lock);
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+}
+
+/** @brief Find next packet
+ * @return Packet to play or NULL if none found
+ *
+ * The return packet is merely guaranteed not to be in the past: it might be
+ * the first packet in the future rather than one that is actually suitable to
+ * play.
+ *
+ * Must be called with @ref lock held.
+ */
+static struct packet *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;
+ }
+ return 0;
+}
+
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
/** @brief Callback from Core Audio */
static OSStatus adioproc
void attribute((unused)) *inClientData) {
UInt32 nbuffers = outOutputData->mNumberBuffers;
AudioBuffer *ab = outOutputData->mBuffers;
- const struct packet *p;
uint32_t samples_available;
- struct timeval in, out;
- gettimeofday(&in, 0);
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;
+ const struct packet *p = next_packet();
if(p && contains(p, next_timestamp)) {
if(p->flags & IDLE)
- fprintf(stderr, "\nIDLE\n");
+ write(2, "I", 1);
/* 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));
+ const uint16_t *ptr = (void *)(p->samples_raw + offset);
samples_available = packet_end - next_timestamp;
if(samples_available > samplesOutLeft)
--nbuffers;
}
pthread_mutex_unlock(&lock);
- gettimeofday(&out, 0);
- {
- static double max;
- double thistime = (out.tv_sec - in.tv_sec) + (out.tv_usec - in.tv_usec) / 1000000.0;
- if(thistime > max)
- fprintf(stderr, "adioproc: %8.8fs\n", max = thistime);
- }
return 0;
}
#endif
+
+#if API_ALSA
+/** @brief PCM handle */
+static snd_pcm_t *pcm;
+
+/** @brief True when @ref pcm is up and running */
+static int alsa_prepared = 1;
+
+/** @brief Initialize @ref pcm */
+static void setup_alsa(void) {
+ 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;
+ int err;
+
+ /* Open ALSA */
+ if((err = snd_pcm_open(&pcm,
+ device ? device : "default",
+ SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK)))
+ fatal(0, "error from snd_pcm_open: %d", err);
+ /* Set up 'hardware' parameters */
+ snd_pcm_hw_params_alloca(&hwparams);
+ if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_any: %d", err);
+ if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
+ if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
+ sample_format)) < 0)
+
+ fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
+ sample_format, err);
+ if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
+ rate, err);
+ if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
+ channels)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
+ channels, err);
+ if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
+ &pcm_bufsize)) < 0)
+ fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
+ MAXSAMPLES * samplesize * 3, err);
+ if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
+ fatal(0, "error calling snd_pcm_hw_params: %d", err);
+ /* Set up 'software' parameters */
+ snd_pcm_sw_params_alloca(&swparams);
+ if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
+ fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
+ if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
+ fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
+ avail_min, err);
+ if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
+ fatal(0, "error calling snd_pcm_sw_params: %d", err);
+}
+
+/** @brief Wait until ALSA wants some audio */
+static void wait_alsa(void) {
+ struct pollfd fds[64];
+ int nfds, err;
+ unsigned short events;
+
+ for(;;) {
+ do {
+ if((nfds = snd_pcm_poll_descriptors(pcm,
+ fds, sizeof fds / sizeof *fds)) < 0)
+ fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds);
+ } while(poll(fds, nfds, -1) < 0 && errno == EINTR);
+ if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events)))
+ fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
+ if(events & POLLOUT)
+ return;
+ }
+}
+
+/** @brief Play some sound via ALSA
+ * @param s Pointer to sample data
+ * @param n Number of samples
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_writei(const void *s, size_t n) {
+ /* Do the write */
+ const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2);
+ if(frames_written < 0) {
+ /* Something went wrong */
+ switch(frames_written) {
+ case -EAGAIN:
+ write(2, "#", 1);
+ return 0;
+ case -EPIPE:
+ error(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ return -1;
+ default:
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ }
+ } else {
+ /* Success */
+ next_timestamp += frames_written * 2;
+ return 0;
+ }
+}
+
+/** @brief Play the relevant part of a packet
+ * @param p Packet to play
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_play(const struct packet *p) {
+ if(p->flags & IDLE)
+ write(2, "I", 1);
+ write(2, ".", 1);
+ return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
+ (p->timestamp + p->nsamples) - next_timestamp);
+}
+
+/** @brief Play some silence
+ * @param p Next packet or NULL
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_infill(const struct packet *p) {
+ static const uint16_t zeros[INFILL_SAMPLES];
+ size_t samples_available = INFILL_SAMPLES;
+
+ if(p && samples_available > p->timestamp - next_timestamp)
+ samples_available = p->timestamp - next_timestamp;
+ write(2, "?", 1);
+ return alsa_writei(zeros, samples_available);
+}
+
+/** @brief Reset ALSA state after we lost synchronization */
+static void alsa_reset(int hard_reset) {
+ int err;
+
+ if((err = snd_pcm_nonblock(pcm, 0)))
+ fatal(0, "error calling snd_pcm_nonblock: %d", err);
+ if(hard_reset) {
+ if((err = snd_pcm_drop(pcm)))
+ fatal(0, "error calling snd_pcm_drop: %d", err);
+ } 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);
+ alsa_prepared = 0;
+}
+#endif
+
/** @brief Play an RTP stream
*
* This is the guts of the program. It is responsible for:
pthread_create(<id, 0, listen_thread, 0);
#if API_ALSA
{
- snd_pcm_t *pcm;
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- /* Only support one format for now */
- const int sample_format = SND_PCM_FORMAT_S16_BE;
- unsigned rate = 44100;
- const int channels = 2;
- const int samplesize = channels * sizeof(uint16_t);
- snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
- /* If we can write more than this many samples we'll get a wakeup */
- const int avail_min = 256;
- snd_pcm_sframes_t frames_written;
- size_t samples_written;
- int prepared = 1;
- int err;
- 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);
+ struct packet *p;
+ int escape, err;
+
+ /* Open the sound device */
+ setup_alsa();
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) {
+ fill_buffer();
+ if(!alsa_prepared) {
if((err = snd_pcm_prepare(pcm)))
fatal(0, "error calling snd_pcm_prepare: %d", err);
- prepared = 1;
+ alsa_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 */
+ /* Keep playing until the buffer empties out, or ALSA tells us to get
+ * lost */
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);
- }
+ wait_alsa();
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;
- }
- }
+ /* ALSA is ready for more data, find something to play */
+ p = next_packet();
+ /* Play it or play some silence */
+ if(contains(p, next_timestamp))
+ escape = alsa_play(p);
+ else
+ escape = alsa_infill(p);
}
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;
+ alsa_reset(escape);
pthread_mutex_lock(&lock);
}
pthread_mutex_lock(&lock);
for(;;) {
/* Wait for the buffer to fill up a bit */
- info("Buffering...");
- while(nsamples < readahead)
- pthread_cond_wait(&cond, &lock);
+ fill_buffer();
/* Start playing now */
info("Playing...");
next_timestamp = pheap_first(&packets)->timestamp;
~mdw / disorder / blobdiff