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De-globalise input_bytes_per_frame,output_bytes_per_frame,output_sample_ratio,max_frame_size_change,decoder_info,shutdown_requested,connection_state_to_output,player_thread_please_stop

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This commit is contained in:
Mike Brady
2017-03-09 16:10:39 +00:00
parent 600cebacd4
commit 05305e1581
3 changed files with 68 additions and 124 deletions
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player.c
+59 -123
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@@ -80,27 +80,13 @@ static uint32_t timestamp_epoch, last_timestamp,
maximum_timestamp_interval; // timestamp_epoch of zero means not initialised, could start at 2
// or 1.
static int input_bytes_per_frame = 4;
static int output_bytes_per_frame;
static int output_sample_ratio;
static int output_rate;
// The maximum frame size change there can be is +/- 1;
static int max_frame_size_change;
// #define FRAME_BYTES(max_frames_per_packet) (4 * max_frames_per_packet)
// maximal resampling shift - conservative
//#define OUTFRAME_BYTES(max_frames_per_packet) (4 * (max_frames_per_packet + 3))
// static pthread_t player_thread = NULL;
static int please_stop;
static int
connection_state_to_output; // if true, then play incoming stuff; if false drop everything
static alac_file *decoder_info;
// debug variables
static int late_packet_message_sent;
static int32_t last_seqno_read;
static int decoder_in_use = 0;
@@ -126,7 +112,6 @@ static int64_t first_packet_timestamp = 0;
static int flush_requested = 0;
static int64_t flush_rtp_timestamp;
static uint64_t time_of_last_audio_packet;
static int shutdown_requested;
// mutexes and condition variables
static pthread_mutex_t ab_mutex = PTHREAD_MUTEX_INITIALIZER;
@@ -294,7 +279,7 @@ static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_co
unsigned char packetp[MAX_PACKET];
assert(len <= MAX_PACKET);
int reply = 0; // everything okay
int outsize = input_bytes_per_frame * (*destlen); // the size the output should be, in bytes
int outsize = conn->input_bytes_per_frame * (*destlen); // the size the output should be, in bytes
int toutsize = outsize;
if (conn->stream.encrypted) {
@@ -326,7 +311,7 @@ static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_co
debug(1, "Hammerton Decoder used on encrypted audio.");
decoder_in_use = 1 << decoder_hammerton;
}
alac_decode_frame(decoder_info, packet, (unsigned char *)dest, &outsize);
alac_decode_frame(conn->decoder_info, packet, (unsigned char *)dest, &outsize);
}
} else {
// not encrypted
@@ -345,7 +330,7 @@ static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_co
debug(1, "Hammerton Decoder used on unencrypted audio.");
decoder_in_use = 1 << decoder_hammerton;
}
alac_decode_frame(decoder_info, buf, dest, &outsize);
alac_decode_frame(conn->decoder_info, buf, dest, &outsize);
}
}
@@ -356,11 +341,11 @@ static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_co
reply = -1; // output packet is the wrong size
}
*destlen = outsize / input_bytes_per_frame;
if ((outsize % input_bytes_per_frame) != 0)
*destlen = outsize / conn->input_bytes_per_frame;
if ((outsize % conn->input_bytes_per_frame) != 0)
debug(1, "Number of audio frames (%d) does not correspond exactly to the number of bytes (%d) "
"and the audio frame size (%d).",
*destlen, outsize, input_bytes_per_frame);
*destlen, outsize, conn->input_bytes_per_frame);
return reply;
}
@@ -451,12 +436,12 @@ static int init_decoder(int32_t fmtp[12],rtsp_conn_info* conn) {
conn->input_num_channels = fmtp[7];
conn->input_bit_depth = fmtp[3];
input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
alac = alac_create(conn->input_bit_depth, conn->input_num_channels);
if (!alac)
return 1;
decoder_info = alac;
conn->decoder_info = alac;
alac->setinfo_max_samples_per_frame = conn->max_frames_per_packet;
alac->setinfo_7a = fmtp[2];
@@ -478,8 +463,8 @@ static int init_decoder(int32_t fmtp[12],rtsp_conn_info* conn) {
return 0;
}
static void terminate_decoders(void) {
alac_free(decoder_info);
static void terminate_decoders(rtsp_conn_info* conn) {
alac_free(conn->decoder_info);
#ifdef HAVE_APPLE_ALAC
apple_alac_terminate();
#endif
@@ -489,7 +474,7 @@ static void init_buffer(rtsp_conn_info* conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
conn->audio_buffer[i].data =
malloc(input_bytes_per_frame * (conn->max_frames_per_packet + max_frame_size_change));
malloc(conn->input_bytes_per_frame * (conn->max_frames_per_packet + conn->max_frame_size_change));
ab_resync(conn);
}
@@ -503,7 +488,7 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len, r
// all timestamps are done at the output rate
int64_t ltimestamp = timestamp * output_sample_ratio;
int64_t ltimestamp = timestamp * conn->output_sample_ratio;
// ignore a request to flush that has been made before the first packet...
if (conn->packet_count == 0) {
@@ -516,7 +501,7 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len, r
pthread_mutex_lock(&ab_mutex);
conn->packet_count++;
time_of_last_audio_packet = get_absolute_time_in_fp();
if (connection_state_to_output) { // if we are supposed to be processing these packets
if (conn->connection_state_to_output) { // if we are supposed to be processing these packets
// if (flush_rtp_timestamp != 0)
// debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
@@ -566,13 +551,6 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len, r
abuf = conn->audio_buffer + BUFIDX(seqno);
} else { // too late.
too_late_packets++;
/*
if (!late_packet_message_sent) {
debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read,
ab_write);
late_packet_message_sent=1;
}
*/
}
// pthread_mutex_unlock(&ab_mutex);
@@ -730,41 +708,6 @@ static inline void process_sample(int32_t sample, char **outp, enum sps_format_t
*outp += result;
}
static inline short dithered_vol(short sample) {
long out;
out = (long)sample * fix_volume;
if (fix_volume < 0x10000) {
// add a TPDF dither -- see
// http://www.users.qwest.net/%7Evolt42/cadenzarecording/DitherExplained.pdf
// and the discussion around https://www.hydrogenaud.io/forums/index.php?showtopic=16963&st=25
// I think, for a 32 --> 16 bits, the range of
// random numbers needs to be from -2^16 to 2^16, i.e. from -65536 to 65536 inclusive, not from
// -32768 to +32767
// See the original paper at
// http://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lipshitz_1992.pdf
// by Lipshitz, Wannamaker and Vanderkooy, 1992.
long tpdf = rand_in_range(65536 + 1) - rand_in_range(65536 + 1);
// Check there's no clipping -- if there is,
if (tpdf >= 0) {
if (LONG_MAX - tpdf >= out)
out += tpdf;
else
out = LONG_MAX;
} else {
if (LONG_MIN - tpdf <= out)
out += tpdf;
else
out = LONG_MIN;
}
}
return out >> 16;
}
// get the next frame, when available. return 0 if underrun/stream reset.
static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
int16_t buf_fill;
@@ -787,22 +730,22 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
// config.timeout of zero means don't check..., but iTunes may be confused by a long gap
// followed by a resumption...
if ((time_of_last_audio_packet != 0) && (shutdown_requested == 0) &&
if ((time_of_last_audio_packet != 0) && (conn->shutdown_requested == 0) &&
(config.dont_check_timeout == 0)) {
uint64_t ct = config.timeout; // go from int to 64-bit int
if ((local_time_now > time_of_last_audio_packet) &&
(local_time_now - time_of_last_audio_packet >= ct << 32)) {
debug(1, "As Yeats almost said, \"Too long a silence / can make a stone of the heart\"");
rtsp_request_shutdown_stream();
shutdown_requested = 1;
conn->shutdown_requested = 1;
}
}
int rco = get_requested_connection_state_to_output();
if (connection_state_to_output != rco) {
connection_state_to_output = rco;
if (conn->connection_state_to_output != rco) {
conn->connection_state_to_output = rco;
// change happening
if (connection_state_to_output == 0) { // going off
if (conn->connection_state_to_output == 0) { // going off
pthread_mutex_lock(&flush_mutex);
flush_requested = 1;
pthread_mutex_unlock(&flush_mutex);
@@ -870,7 +813,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
uint64_t reference_timestamp_time, remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
&remote_reference_timestamp_time);
reference_timestamp *= output_sample_ratio;
reference_timestamp *= conn->output_sample_ratio;
if (first_packet_timestamp == 0) { // if this is the very first packet
// debug(1,"First frame seen, time %u, with %d
// frames...",curframe->timestamp,seq_diff(ab_read, ab_write));
@@ -903,10 +846,10 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
// if would be in sync. To do this, we would give it a latency offset of -100 ms, i.e.
// -4410 frames.
debug(1, "Output sample ratio is %d", output_sample_ratio);
debug(1, "Output sample ratio is %d", conn->output_sample_ratio);
int64_t delta = (first_packet_timestamp - reference_timestamp) +
config.latency * output_sample_ratio +
config.latency * conn->output_sample_ratio +
config.audio_backend_latency_offset * config.output_rate;
if (delta >= 0) {
@@ -924,7 +867,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
debug(
1,
"First packet is late! It should have played before now. Flushing 0.1 seconds");
player_flush(first_packet_timestamp + 4410 * output_sample_ratio);
player_flush(first_packet_timestamp + 4410 * conn->output_sample_ratio);
}
}
}
@@ -932,7 +875,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
if (first_packet_time_to_play != 0) {
// recalculate first_packet_time_to_play -- the latency might change
int64_t delta = (first_packet_timestamp - reference_timestamp) +
config.latency * output_sample_ratio +
config.latency * conn->output_sample_ratio +
config.audio_backend_latency_offset * config.output_rate;
if (delta >= 0) {
@@ -1011,11 +954,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
// ouotputting frames for a while -- it could get loaded up but not start responding
// for many milliseconds.
if (fs != 0) {
silence = malloc(output_bytes_per_frame * fs);
silence = malloc(conn->output_bytes_per_frame * fs);
if (silence == NULL)
debug(1, "Failed to allocate %d byte silence buffer.", fs);
else {
memset(silence, 0, output_bytes_per_frame * fs);
memset(silence, 0, conn->output_bytes_per_frame * fs);
// debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
// with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write));
config.output->play(silence, fs);
@@ -1031,7 +974,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
uint64_t reference_timestamp_time; // don't need this...
get_reference_timestamp_stuff(&play_segment_reference_frame, &reference_timestamp_time,
&play_segment_reference_frame_remote_time);
play_segment_reference_frame *= output_sample_ratio;
play_segment_reference_frame *= conn->output_sample_ratio;
#ifdef CONFIG_METADATA
send_ssnc_metadata('prsm', NULL, 0,
0); // "resume", but don't wait if the queue is locked
@@ -1061,12 +1004,12 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
uint64_t reference_timestamp_time, remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
&remote_reference_timestamp_time); // all types okay
reference_timestamp *= output_sample_ratio;
reference_timestamp *= conn->output_sample_ratio;
if (reference_timestamp) { // if we have a reference time
int64_t packet_timestamp = curframe->timestamp; // types okay
int64_t delta = packet_timestamp - reference_timestamp;
int64_t offset =
config.latency * output_sample_ratio +
config.latency * conn->output_sample_ratio +
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
@@ -1098,7 +1041,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
}
do_wait = 1;
}
wait = (ab_buffering || (do_wait != 0) || (!ab_synced)) && (!please_stop);
wait = (ab_buffering || (do_wait != 0) || (!ab_synced)) && (!conn->player_thread_please_stop);
if (wait) {
uint64_t time_to_wait_for_wakeup_fp =
@@ -1131,7 +1074,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
}
} while (wait);
if (please_stop) {
if (conn->player_thread_please_stop) {
pthread_mutex_unlock(&ab_mutex);
return 0;
}
@@ -1313,10 +1256,6 @@ static int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int leng
for (i = 0; i < length + tstuff; i++) {
process_sample(*ip++, &l_outptr, l_output_format, fix_volume, dither,conn);
process_sample(*ip++, &l_outptr, l_output_format, fix_volume, dither,conn);
//*op = dithered_vol(*op);
// op++;
//*op = dithered_vol(*op);
// op++;
};
} else { // the whole frame, if no stuffing
@@ -1329,8 +1268,6 @@ static int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int leng
for (i = 0; i < length; i++) {
process_sample(*ip++, &l_outptr, l_output_format, fix_volume, dither,conn);
process_sample(*ip++, &l_outptr, l_output_format, fix_volume, dither,conn);
//*op++ = dithered_vol(*ip++);
//*op++ = dithered_vol(*ip++);
};
}
return length + tstuff;
@@ -1348,13 +1285,14 @@ static void *player_thread_func(void *arg) {
conn->please_stop = 0;
conn->packet_count = 0;
conn->previous_random_number = 0;
conn->input_bytes_per_frame = 4;
conn->player_thread_please_stop = 0;
init_decoder((int32_t *)&conn->stream.fmtp,conn); // this sets up incoming rate, bit depth, channels
// must be after decoder init
init_buffer(conn);
debug(1, "Output frame bytes is %d.", output_bytes_per_frame);
debug(1, "Output frame bytes is %d.", conn->output_bytes_per_frame);
if (conn->stream.encrypted) {
#ifdef HAVE_LIBMBEDTLS
@@ -1377,24 +1315,24 @@ static void *player_thread_func(void *arg) {
// seconds of a gap between successive rtptimes, at
// worst
output_sample_ratio = config.output_rate / conn->input_rate;
conn->output_sample_ratio = config.output_rate / conn->input_rate;
debug(1, "Output sample ratio is %d.", output_sample_ratio);
debug(1, "Output sample ratio is %d.", conn->output_sample_ratio);
max_frame_size_change = 1 * output_sample_ratio; // we add or subtract one frame at the nominal
conn->max_frame_size_change = 1 * conn->output_sample_ratio; // we add or subtract one frame at the nominal
// rate, multiply it by the frame ratio.
output_bytes_per_frame = 4;
conn->output_bytes_per_frame = 4;
switch (config.output_format) {
case SPS_FORMAT_S24_3LE:
case SPS_FORMAT_S24_3BE:
output_bytes_per_frame = 6;
conn->output_bytes_per_frame = 6;
break;
case SPS_FORMAT_S24:
output_bytes_per_frame = 8;
conn->output_bytes_per_frame = 8;
break;
case SPS_FORMAT_S32:
output_bytes_per_frame = 8;
conn->output_bytes_per_frame = 8;
break;
}
@@ -1414,7 +1352,7 @@ static void *player_thread_func(void *arg) {
die("Not enough buffers available for a total latency of %d frames. A maximum of %d 352-frame "
"packets may be accommodated.",
maximum_latency, BUFFER_FRAMES);
connection_state_to_output = get_requested_connection_state_to_output();
conn->connection_state_to_output = get_requested_connection_state_to_output();
// this is about half a minute
#define trend_interval 3758
stats_t statistics[trend_interval];
@@ -1435,7 +1373,7 @@ static void *player_thread_func(void *arg) {
int64_t current_delay;
int play_number = 0;
time_of_last_audio_packet = 0;
shutdown_requested = 0;
conn->shutdown_requested = 0;
number_of_statistics = oldest_statistic = newest_statistic = 0;
tsum_of_sync_errors = tsum_of_corrections = tsum_of_insertions_and_deletions = tsum_of_drifts = 0;
@@ -1490,13 +1428,13 @@ static void *player_thread_func(void *arg) {
// debug(1,"Define tbuf of length
// %d.",output_bytes_per_frame*(max_frames_per_packet*output_sample_ratio+max_frame_size_change));
tbuf = malloc(sizeof(int32_t) * 2 *
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
(conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
if (tbuf == NULL)
debug(1, "Failed to allocate memory for the transition buffer.");
sbuf = 0;
if (config.packet_stuffing == ST_soxr) { // needed for stuffing
sbuf = malloc(sizeof(int32_t) * 2 *
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
(conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
if (sbuf == NULL)
debug(1, "Failed to allocate memory for the transition buffer.");
}
@@ -1504,15 +1442,14 @@ static void *player_thread_func(void *arg) {
// We might need an output buffer and a buffer of silence.
// The size of these dependents on the number of frames, the size of each frame and the maximum
// size change
outbuf = malloc(output_bytes_per_frame *
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
outbuf = malloc(conn->output_bytes_per_frame *
(conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
if (outbuf == NULL)
debug(1, "Failed to allocate memory for an output buffer.");
silence = malloc(output_bytes_per_frame * conn->max_frames_per_packet * output_sample_ratio);
silence = malloc(conn->output_bytes_per_frame * conn->max_frames_per_packet * conn->output_sample_ratio);
if (silence == NULL)
debug(1, "Failed to allocate memory for a silence buffer.");
memset(silence, 0, output_bytes_per_frame * conn->max_frames_per_packet * output_sample_ratio);
late_packet_message_sent = 0;
memset(silence, 0, conn->output_bytes_per_frame * conn->max_frames_per_packet * conn->output_sample_ratio);
first_packet_timestamp = 0;
missing_packets = late_packets = too_late_packets = resend_requests = 0;
flush_rtp_timestamp = 0; // it seems this number has a special significance -- it seems to be used
@@ -1558,7 +1495,7 @@ static void *player_thread_func(void *arg) {
}
uint64_t tens_of_seconds = 0;
while (!please_stop) {
while (!conn->player_thread_please_stop) {
abuf_t *inframe = buffer_get_frame(conn);
if (inframe) {
inbuf = inframe->data;
@@ -1569,7 +1506,7 @@ static void *player_thread_func(void *arg) {
// debug(1,"Player has a supplied silent frame.");
last_seqno_read =
(SUCCESSOR(last_seqno_read) & 0xffff); // manage the packet out of sequence minder
config.output->play(silence, conn->max_frames_per_packet * output_sample_ratio);
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
} else {
int enable_dither = 0;
@@ -1620,7 +1557,7 @@ static void *player_thread_func(void *arg) {
// here, replicate the samples if you're upsampling
for (j = 0; j < output_sample_ratio; j++) {
for (j = 0; j < conn->output_sample_ratio; j++) {
// raise the 16-bit sample to 32 bits.
int32_t t = ls << 16;
*outpl++ = t;
@@ -1656,7 +1593,7 @@ static void *player_thread_func(void *arg) {
}
// inbuf = tbuf;
inbuflength *= output_sample_ratio;
inbuflength *= conn->output_sample_ratio;
}
// We have a frame of data. We need to see if we want to add or remove a frame from it to
@@ -1672,7 +1609,7 @@ static void *player_thread_func(void *arg) {
uint64_t reference_timestamp_time, remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
&remote_reference_timestamp_time); // types okay
reference_timestamp *= output_sample_ratio;
reference_timestamp *= conn->output_sample_ratio;
int64_t rt, nt;
rt = reference_timestamp; // uint32_t to int64_t
nt = inframe->timestamp; // uint32_t to int64_t
@@ -1766,7 +1703,7 @@ static void *player_thread_func(void *arg) {
// This is the timing error for the next audio frame in the DAC.
sync_error =
delay -
config.latency * output_sample_ratio; // int64_t from int64_t - int32_t, so okay
config.latency * conn->output_sample_ratio; // int64_t from int64_t - int32_t, so okay
// if (llabs(sync_error)>352*512)
// debug(1,"Very large sync error: %lld",sync_error);
@@ -1865,7 +1802,7 @@ static void *player_thread_func(void *arg) {
if (abs_sync_error < 0)
abs_sync_error = -abs_sync_error;
if ((config.no_sync == 0) && (inframe->timestamp != 0) && (!please_stop) &&
if ((config.no_sync == 0) && (inframe->timestamp != 0) && (!conn->player_thread_please_stop) &&
(config.resyncthreshold > 0.0) &&
(abs_sync_error > config.resyncthreshold * config.output_rate)) {
sync_error_out_of_bounds++;
@@ -1992,9 +1929,9 @@ static void *player_thread_func(void *arg) {
"%*d", /* max buffer occupancy */
10,
1000 * moving_average_sync_error / config.output_rate, 10,
moving_average_correction * 1000000 / (352 * output_sample_ratio), 10,
moving_average_correction * 1000000 / (352 * conn->output_sample_ratio), 10,
moving_average_insertions_plus_deletions * 1000000 /
(352 * output_sample_ratio),
(352 * conn->output_sample_ratio),
12, play_number, 7, missing_packets, 7, late_packets, 7, too_late_packets,
7, resend_requests, 7, minimum_dac_queue_size, 5, minimum_buffer_occupancy,
5, maximum_buffer_occupancy);
@@ -2072,7 +2009,7 @@ static void *player_thread_func(void *arg) {
pthread_join(rtp_control_thread, NULL);
debug(1, "control thread joined");
free_buffer(conn);
terminate_decoders();
terminate_decoders(conn);
debug(1, "Player thread exit");
return 0;
}
@@ -2290,7 +2227,6 @@ int player_play(pthread_t *player_thread, rtsp_conn_info *conn) {
if (config.buffer_start_fill > BUFFER_FRAMES)
die("specified buffer starting fill %d > buffer size %d", config.buffer_start_fill,
BUFFER_FRAMES);
please_stop = 0;
command_start();
#ifdef CONFIG_METADATA
send_ssnc_metadata('pbeg', NULL, 0, 1);
@@ -2324,7 +2260,7 @@ void player_stop(pthread_t *player_thread, rtsp_conn_info *conn) {
// if (*thread==NULL)
// debug(1,"Trying to stop a non-existent player thread");
// else {
please_stop = 1;
conn->player_thread_please_stop = 1;
pthread_cond_signal(&flowcontrol); // tell it to give up
pthread_join(*player_thread, NULL);
#ifdef CONFIG_METADATA
player.h
+8
View File
@@ -18,6 +18,7 @@
#endif
#include "audio.h"
#include "alac.h"
typedef uint16_t seq_t;
@@ -51,9 +52,16 @@ typedef struct {
abuf_t audio_buffer[BUFFER_FRAMES];
int max_frames_per_packet,input_num_channels,input_bit_depth,input_rate;
int input_bytes_per_frame,output_bytes_per_frame,output_sample_ratio;
int max_frame_size_change;
int64_t previous_random_number;
alac_file *decoder_info;
uint32_t please_stop;
uint64_t packet_count;
int shutdown_requested;
int connection_state_to_output;
int player_thread_please_stop;
#ifdef HAVE_LIBMBEDTLS
mbedtls_aes_context dctx;
#endif
rtp.c
+1 -1
View File
@@ -129,7 +129,7 @@ void *rtp_audio_receiver(void *arg) {
stat_mean += stat_delta / stat_n;
stat_M2 += stat_delta * (time_interval_us - stat_mean);
if (stat_n % 2500 == 0) {
debug(2, "Packet reception interval stats: mean, standard deviation and max for the last "
debug(3, "Packet reception interval stats: mean, standard deviation and max for the last "
"2,500 packets in microseconds: %10.1f, %10.1f, %10.1f.",
stat_mean, sqrtf(stat_M2 / (stat_n - 1)), longest_packet_time_interval_us);
stat_n = 0;