/** @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>
/** @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
*
/** @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;
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;
--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
}
}
-/** @brief Play some sound
+/** @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
/* Something went wrong */
switch(frames_written) {
case -EAGAIN:
+ write(2, "#", 1);
return 0;
case -EPIPE:
error(0, "error calling snd_pcm_writei: %ld",
* @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);
}
#endif
-/** @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;
-}
-
/** @brief Play an RTP stream
*
* This is the guts of the program. It is responsible for:
~mdw / disorder / blobdiff