[player] Fix for player getting ahead of network streams (issue #218)

Issue caused stuttering after a few hours. This fix will check if
the player is getting behind the playback timer. Apparently, we
sometimes get samples at a rate slightly below 44100 from network
streams, and then we end up consuming too quickly. This introduces
a way of reducing consumption if that happens.
This commit is contained in:
ejurgensen 2016-02-11 08:29:29 +01:00
parent 19b69a4f67
commit 5f5a138c77
1 changed files with 79 additions and 56 deletions

View File

@ -75,6 +75,10 @@
// Default volume (must be from 0 - 100)
#define PLAYER_DEFAULT_VOLUME 50
// Used to keep the player from getting ahead of a rate limited source (see below)
#define PLAYER_TICKS_MAX_DELAY 2
// Skips ticks for about 2 secs (seems to bring us back in sync for about 20 min)
#define PLAYER_TICKS_SKIP 126
struct player_source
{
@ -259,9 +263,15 @@ timer_t pb_timer;
static struct event *pb_timer_ev;
static struct timespec pb_timer_last;
static struct timespec packet_timer_last;
static uint64_t MINIMUM_STREAM_PERIOD;
static struct timespec packet_time = { 0, AIRTUNES_V2_STREAM_PERIOD };
// How often the playback timer triggers player_playback_cb
static struct timespec tick_interval;
// Timer resolution
static struct timespec timer_res;
// Time between two packets
static struct timespec packet_time = { 0, AIRTUNES_V2_STREAM_PERIOD };
// Will be positive if we need to skip some source reads (see below)
static int ticks_skip;
/* Sync source */
static enum player_sync_source pb_sync_source;
@ -555,41 +565,25 @@ player_get_current_pos(uint64_t *pos, struct timespec *ts, int commit)
}
static int
pb_timer_start(struct timespec *ts)
pb_timer_start(void)
{
struct itimerspec next;
struct itimerspec tick;
int ret;
next.it_interval.tv_sec = 0;
next.it_interval.tv_nsec = 0;
next.it_value.tv_sec = ts->tv_sec;
next.it_value.tv_nsec = ts->tv_nsec;
tick.it_interval = tick_interval;
tick.it_value = tick_interval;
#if defined(__linux__)
ret = timerfd_settime(pb_timer_fd, TFD_TIMER_ABSTIME, &next, NULL);
if (ret < 0)
{
DPRINTF(E_LOG, L_PLAYER, "Could not arm playback timer: %s\n", strerror(errno));
return -1;
}
ret = event_add(pb_timer_ev, NULL);
if (ret < 0)
{
DPRINTF(E_LOG, L_PLAYER, "Could not add playback timer event\n");
return -1;
}
ret = timerfd_settime(pb_timer_fd, 0, &tick, NULL);
#else
ret = timer_settime(pb_timer, TIMER_ABSTIME, &next, NULL);
ret = timer_settime(pb_timer, 0, &tick, NULL);
#endif
if (ret < 0)
{
DPRINTF(E_LOG, L_PLAYER, "Could not arm playback timer: %s\n", strerror(errno));
return -1;
}
#endif
return 0;
}
@ -597,17 +591,15 @@ pb_timer_start(struct timespec *ts)
static int
pb_timer_stop(void)
{
struct itimerspec next;
struct itimerspec tick;
int ret;
memset(&next, 0, sizeof(struct itimerspec));
memset(&tick, 0, sizeof(struct itimerspec));
#if defined(__linux__)
ret = timerfd_settime(pb_timer_fd, TFD_TIMER_ABSTIME, &next, NULL);
event_del(pb_timer_ev);
ret = timerfd_settime(pb_timer_fd, 0, &tick, NULL);
#else
ret = timer_settime(pb_timer, TIMER_ABSTIME, &next, NULL);
ret = timer_settime(pb_timer, 0, &tick, NULL);
#endif
if (ret < 0)
{
@ -1620,24 +1612,59 @@ playback_write(void)
static void
player_playback_cb(int fd, short what, void *arg)
{
uint32_t packet_send_count = 0;
struct timespec next_tick;
struct timespec stream_period = { 0, MINIMUM_STREAM_PERIOD };
int ret;
#if defined(__linux__)
uint64_t ticks;
uint32_t packet_send_count;
int ret;
/* Acknowledge timer */
// Check if we missed any timer expirations
ticks = 0;
#if defined(__linux__)
ret = read(fd, &ticks, sizeof(ticks));
if (ret <= 0)
DPRINTF(E_WARN, L_PLAYER, "Error reading timer.\n");
DPRINTF(E_LOG, L_PLAYER, "Error reading timer\n");
else
ticks--;
#else
ret = timer_getoverrun(pb_timer);
if (ret < 0)
DPRINTF(E_LOG, L_PLAYER, "Error getting timer overrun\n");
else
ticks = ret;
#endif /* __linux__ */
/* Decide how many packets to send */
next_tick = timespec_add(pb_timer_last, stream_period);
// The reason we get behind the playback timer may be that we are playing a
// network stream. We might be consuming faster than the stream delivers, so
// when ffmpeg's buffer empties (might take a few hours) our av_read_frame()
// in transcode.c will begin to block, because ffmpeg has to wait for new data
// from the stream server. In that situation we will skip reading data every
// second tick until we have skipt PLAYER_TICKS_SKIP ticks. That should make
// the source catch up. RTP destinations should be able to handle this
// gracefully if we just give them an rtptime that lets them know that some
// packets were "lost".
if (ticks > PLAYER_TICKS_MAX_DELAY)
{
DPRINTF(E_WARN, L_PLAYER, "Behind the playback timer with %" PRIu64 " ticks, initiating catch up\n", ticks);
ticks_skip = 2 * PLAYER_TICKS_SKIP + 1;
}
else if (ticks_skip > 0)
ticks_skip--;
// Decide how many packets to send
next_tick = timespec_add(pb_timer_last, tick_interval);
for (; ticks > 0; ticks--)
next_tick = timespec_add(next_tick, tick_interval);
packet_send_count = 0;
do
{
// Skip reading and writing every second tick if we are behind the source
if (ticks_skip % 2 == 0)
playback_write();
else
last_rtptime += AIRTUNES_V2_PACKET_SAMPLES;
packet_timer_last = timespec_add(packet_timer_last, packet_time);
packet_send_count++;
/* not possible to have more than 126 audio packets per second */
@ -1655,11 +1682,7 @@ player_playback_cb(int fd, short what, void *arg)
if (player_state == PLAY_STOPPED)
return;
pb_timer_last = timespec_add(pb_timer_last, stream_period);
ret = pb_timer_start(&pb_timer_last);
if (ret < 0)
playback_abort();
pb_timer_last = next_tick;
}
static void
@ -2407,7 +2430,7 @@ playback_start_bh(struct player_command *cmd)
pb_timer_last.tv_sec = pb_pos_stamp.tv_sec;
pb_timer_last.tv_nsec = pb_pos_stamp.tv_nsec;
ret = pb_timer_start(&pb_timer_last);
ret = pb_timer_start();
if (ret < 0)
goto out_fail;
@ -4750,6 +4773,7 @@ exit_cb(int fd, short what, void *arg)
int
player_init(void)
{
uint64_t interval;
uint32_t rnd;
int raop_v6enabled;
int mdns_flags;
@ -4800,11 +4824,13 @@ player_init(void)
timer_res.tv_nsec = 2 * AIRTUNES_V2_STREAM_PERIOD;
#endif
MINIMUM_STREAM_PERIOD = MAX(timer_res.tv_nsec, AIRTUNES_V2_STREAM_PERIOD);
// Set the tick interval for the playback timer
interval = MAX(timer_res.tv_nsec, AIRTUNES_V2_STREAM_PERIOD);
tick_interval.tv_nsec = interval;
/* Create a timer */
// Create the playback timer
#if defined(__linux__)
pb_timer_fd = timerfd_create(CLOCK_MONOTONIC, TFD_CLOEXEC);
pb_timer_fd = timerfd_create(CLOCK_MONOTONIC, TFD_CLOEXEC | TFD_NONBLOCK);
ret = pb_timer_fd;
#else
ret = timer_create(CLOCK_MONOTONIC, NULL, &pb_timer);
@ -4816,7 +4842,7 @@ player_init(void)
return -1;
}
/* Random RTP time start */
// Random RTP time start
gcry_randomize(&rnd, sizeof(rnd), GCRY_STRONG_RANDOM);
last_rtptime = ((uint64_t)1 << 32) | rnd;
@ -4824,7 +4850,7 @@ player_init(void)
if (ret < 0)
laudio_volume = PLAYER_DEFAULT_VOLUME;
else if (laudio_selected)
speaker_select_laudio(); /* Run the select helper */
speaker_select_laudio(); // Run the select helper
audio_buf = evbuffer_new();
if (!audio_buf)
@ -4875,9 +4901,9 @@ player_init(void)
}
#if defined(__linux__)
pb_timer_ev = event_new(evbase_player, pb_timer_fd, EV_READ, player_playback_cb, NULL);
pb_timer_ev = event_new(evbase_player, pb_timer_fd, EV_READ | EV_PERSIST, player_playback_cb, NULL);
#else
pb_timer_ev = evsignal_new(evbase_player, SIGALRM, player_playback_cb, NULL);
pb_timer_ev = event_new(evbase_player, SIGALRM, EV_SIGNAL | EV_PERSIST, player_playback_cb, NULL);
#endif
if (!pb_timer_ev)
{
@ -4887,10 +4913,7 @@ player_init(void)
event_add(exitev, NULL);
event_add(cmdev, NULL);
#ifndef __linux__
event_add(pb_timer_ev, NULL);
#endif
ret = laudio_init(player_laudio_status_cb);
if (ret < 0)