nils/shairport-sync
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Move a play session's latency into the play session's data structure, duh.

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This commit is contained in:
Mike Brady
2018-01-09 20:54:03 +00:00
parent 9cfc6c475b
commit 3e5dbe9e1a
5 changed files with 15 additions and 16 deletions
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common.h
-1
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@@ -121,7 +121,6 @@ typedef struct {
char *mdns_name;
mdns_backend *mdns;
int buffer_start_fill;
int64_t latency;
int64_t userSuppliedLatency; // overrides all other latencies -- use with caution
int64_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual total latency
// the total latency will be limited to the min and max-latency values, if supplied
player.c
+5 -5
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@@ -862,7 +862,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// debug(1, "Output sample ratio is %d", conn->output_sample_ratio);
int64_t delta = (conn->first_packet_timestamp - reference_timestamp) +
config.latency * conn->output_sample_ratio +
conn->latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate);
if (delta >= 0) {
@@ -888,7 +888,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (conn->first_packet_time_to_play != 0) {
// recalculate conn->first_packet_time_to_play -- the latency might change
int64_t delta = (conn->first_packet_timestamp - reference_timestamp) +
config.latency * conn->output_sample_ratio +
conn->latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate);
if (delta >= 0) {
@@ -1088,7 +1088,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
int64_t packet_timestamp = curframe->timestamp; // types okay
int64_t delta = packet_timestamp - reference_timestamp;
int64_t offset =
config.latency * conn->output_sample_ratio +
conn->latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate) -
config.audio_backend_buffer_desired_length *
config.output_rate; // all arguments are int32_t, so expression promotion okay
@@ -1463,7 +1463,7 @@ static void *player_thread_func(void *arg) {
// check that there are enough buffers to accommodate the desired latency and the latency offset
int maximum_latency =
config.latency + (int)(config.audio_backend_latency_offset * config.output_rate);
conn->latency + (int)(config.audio_backend_latency_offset * config.output_rate);
if ((maximum_latency + (352 - 1)) / 352 + 10 > BUFFER_FRAMES)
die("Not enough buffers available for a total latency of %d frames. A maximum of %d 352-frame "
"packets may be accommodated.",
@@ -1841,7 +1841,7 @@ static void *player_thread_func(void *arg) {
// if negative, the packet will be early -- the delay is less than expected.
sync_error =
delay - (config.latency * conn->output_sample_ratio +
delay - (conn->latency * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset *
config.output_rate)); // int64_t from int64_t - int32_t, so okay
player.h
+1
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@@ -63,6 +63,7 @@ typedef struct {
typedef struct {
int connection_number; // for debug ID purposes, nothing else...
int64_t latency; // the actual latency used for this play session
int64_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero otherwise
int64_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero otherwise
rtp.c
+7 -6
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@@ -1,7 +1,7 @@
/*
* Apple RTP protocol handler. This file is part of Shairport.
* Copyright (c) James Laird 2013
* Copyright (c) Mike Brady 2014 -- 2017
* Copyright (c) Mike Brady 2014 -- 2018
* All rights reserved.
*
* Permission is hereby granted, free of charge, to any person
@@ -225,12 +225,12 @@ void *rtp_control_receiver(void *arg) {
rtp_timestamp_less_latency = monotonic_timestamp(ntohl(*((uint32_t *)&packet[4])), conn);
sync_rtp_timestamp = monotonic_timestamp(ntohl(*((uint32_t *)&packet[16])), conn);
if (config.userSuppliedLatency) {
if (config.userSuppliedLatency != config.latency) {
if (config.userSuppliedLatency != conn->latency) {
debug(1,"Using the user-supplied latency: %lld.",config.userSuppliedLatency);
}
config.latency = config.userSuppliedLatency;
conn->latency = config.userSuppliedLatency;
} else {
int64_t la =
sync_rtp_timestamp - rtp_timestamp_less_latency + config.fixedLatencyOffset;
@@ -239,8 +239,8 @@ void *rtp_control_receiver(void *arg) {
if ((conn->minimum_latency) && (conn->minimum_latency>la))
la = conn->minimum_latency;
if (la != config.latency) {
config.latency = la;
if (la != conn->latency) {
conn->latency = la;
debug(1,"New latency: %lld, sync latency: %lld, minimum latency: %lld, maximum latency: %lld, fixed offset: %lld.",
la,sync_rtp_timestamp - rtp_timestamp_less_latency,conn->minimum_latency,conn->maximum_latency,config.fixedLatencyOffset);
}
@@ -255,6 +255,7 @@ void *rtp_control_receiver(void *arg) {
// it's as if the first sync after a flush or resume is the timing of the next packet
// after the one whose RTP is given. Weird.
}
pthread_mutex_lock(&conn->reference_time_mutex);
conn->remote_reference_timestamp_time = remote_time_of_sync;
conn->reference_timestamp_time =
shairport.c
+2 -4
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@@ -1108,10 +1108,8 @@ int main(int argc, char **argv) {
strcat(configuration_file_path, ".conf");
config.configfile = configuration_file_path;
config.statistics_requested = 0; // don't print stats in the log
config.latency = -1; // -1 means not set. 88200 works well. This is also reset in rtsp.c when play
// is about to start
config.userSuppliedLatency = 0; // zero means none supplied
//config.statistics_requested = 0; // don't print stats in the log
//config.userSuppliedLatency = 0; // zero means none supplied
config.resyncthreshold = 0.05; // 50 ms
config.timeout = 120; // this number of seconds to wait for [more] audio before switching to idle.
config.tolerance =