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Slightly better timing. Bumped version for ejurgensen's new pair_ap work.

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This commit is contained in:
Mike Brady
2021-05-05 18:33:21 +01:00
parent 83b0a84412
commit fd4cc98aba
4 changed files with 22 additions and 19 deletions
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configure.ac
+1 -1
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@@ -2,7 +2,7 @@
# Process this file with autoconf to produce a configure script.
AC_PREREQ([2.50])
AC_INIT([shairport-sync], [4.0d1], [4265913+mikebrady@users.noreply.github.com])
AC_INIT([shairport-sync], [4.0d2], [4265913+mikebrady@users.noreply.github.com])
AM_INIT_AUTOMAKE
AC_CONFIG_SRCDIR([shairport.c])
AC_CONFIG_HEADERS([config.h])
player.c
+10 -8
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@@ -442,6 +442,8 @@ void player_put_packet(int original_format, seq_t seqno, uint32_t actual_timesta
conn->ab_write = seqno;
conn->ab_read = seqno;
conn->ab_synced = 1;
conn->first_packet_timestamp = 0;
debug(1,"synced by first packet");
} else if (original_format == 0) {
// if the packet is coming in original format, the sequence number is important
// otherwise, ignore is by setting it equal to the expected sequence number in ab_write
@@ -1279,7 +1281,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (do_wait == 0)
if ((conn->ab_synced != 0) && (conn->ab_read == conn->ab_write)) { // the buffer is empty!
if (notified_buffer_empty == 0) {
debug(3, "Buffers exhausted.");
debug(1, "Buffers exhausted.");
notified_buffer_empty = 1;
// reset_input_flow_metrics(conn); // don't do a full flush parameters reset
conn->initial_reference_time = 0;
@@ -1559,11 +1561,11 @@ void player_thread_cleanup_handler(void *arg) {
int elapsedMin = (rawSeconds / 60) % 60;
int elapsedSec = rawSeconds % 60;
if (conn->frame_rate_status)
inform("Playback Stopped. Total playing time %02d:%02d:%02d. Input: %0.2f, output: %0.2f "
"frames per second.",
inform("Connection %d: Playback Stopped. Total playing time %02d:%02d:%02d. Input: %0.2f, output: %0.2f "
"frames per second.", conn->connection_number,
elapsedHours, elapsedMin, elapsedSec, conn->input_frame_rate, conn->frame_rate);
else
inform("Playback Stopped. Total playing time %02d:%02d:%02d. Input: %0.2f frames per second.",
inform("Connection %d: Playback Stopped. Total playing time %02d:%02d:%02d. Input: %0.2f frames per second.", conn->connection_number,
elapsedHours, elapsedMin, elapsedSec, conn->input_frame_rate);
}
@@ -2280,7 +2282,7 @@ void *player_thread_func(void *arg) {
int64_t filler_length =
(int64_t)(config.resyncthreshold * config.output_rate); // number of samples
if ((sync_error > 0) && (sync_error > filler_length)) {
debug(2, "Large positive sync error: %" PRId64 ".", sync_error);
debug(1, "Large positive sync error of: %" PRId64 " frames (%f seconds).", sync_error, (sync_error * 1.0)/config.output_rate);
int64_t local_frames_to_drop = sync_error / conn->output_sample_ratio;
uint32_t frames_to_drop_sized = local_frames_to_drop;
@@ -2292,10 +2294,10 @@ void *player_thread_func(void *arg) {
debug_mutex_unlock(&conn->flush_mutex, 3);
} else if ((sync_error < 0) && ((-sync_error) > filler_length)) {
debug(2,
"Large negative sync error: %" PRId64 " with should_be_frame_32 of %" PRIu32
debug(1,
"Large negative sync error of: %" PRId64 " frames (%f seconds), with should_be_frame_32 of %" PRIu32
", nt of %" PRId64 " and current_delay of %" PRId64 ".",
sync_error, should_be_frame_32,
sync_error, (sync_error * 1.0)/config.output_rate, should_be_frame_32,
inframe->given_timestamp * conn->output_sample_ratio, current_delay);
int64_t silence_length = -sync_error;
if (silence_length > (filler_length * 5))
rtp.c
+10 -10
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@@ -290,8 +290,8 @@ void *rtp_control_receiver(void *arg) {
obfp += 2;
};
*obfp = 0;
// get raw timestamp information
// I think that a good way to understand these timestamps is that
// (1) the rtlt below is the timestamp of the frame that should be playing at the
@@ -302,19 +302,19 @@ void *rtp_control_receiver(void *arg) {
// Thus, (3) the latency can be calculated by subtracting the second from the
// first.
// There must be more to it -- there something missing.
// In addition, it seems that if the value of the short represented by the second
// pair of bytes in the packet is 7
// then an extra time lag is expected to be added, presumably by
// the AirPort Express.
// Best guess is that this delay is 11,025 frames.
uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
uint32_t rt = nctohl(&packet[16]); // raw timestamp
uint32_t fl = nctohs(&packet[2]); //
debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and %u,
giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
//debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
@@ -2065,8 +2065,9 @@ void *rtp_buffered_audio_processor(void *arg) {
get_audio_buffer_size_and_occupancy(&player_buffer_size, &player_buffer_occupancy, conn);
// debug(1,"player buffer size and occupancy: %u and %u", player_buffer_size,
// player_buffer_occupancy);
double reqested_lead_time = 0.4; //
if (player_buffer_occupancy >
((0.25 + 0.1) * 44100.0 / 352)) { // must be greater than the lead time.
((reqested_lead_time + 0.1) * 44100.0 / 352)) { // must be greater than the lead time.
// if there is enough stuff in the player's buffer, sleep for a while and try again
// debug(1,"sleep for 20 ms");
usleep(20000); // wait for a while
@@ -2089,8 +2090,7 @@ void *rtp_buffered_audio_processor(void *arg) {
0) {
int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
// debug(1,"lead time in buffered_audio is %f milliseconds.", lead_time * 0.000001);
// ask for 0.5 sec of leadtime
if ((lead_time >= (int64_t)(0.25 * 1000000000)) || (streaming_has_started == 1)) {
if ((lead_time >= (int64_t)(reqested_lead_time * 1000000000)) || (streaming_has_started == 1)) {
if (streaming_has_started == 0)
debug(1, "rtp lead time is %f ms.", 0.000001 * lead_time);
streaming_has_started = 1;
rtsp.c
+1
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@@ -3163,6 +3163,7 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag
#ifdef CONFIG_AIRPLAY_2
conn->airplay_type = ap_1;
conn->timing_type = ts_ntp;
debug(1, "Connection %d. AirPlay 1 Audio Stream Detected.", conn->connection_number);
#endif
conn->stream.type = ast_unknown;