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Merge resolution to issue 694 (buzz during track change) into master branch

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This commit is contained in:
Mike Brady
2018-05-08 20:57:47 +01:00
parent 548d5da997 1ddaa14224
commit 3de38bcd7b
5 changed files with 54 additions and 37 deletions
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audio.h
+1 -1
View File
@@ -22,7 +22,7 @@ typedef struct {
void (*start)(int sample_rate, int sample_format);
// block of samples
void (*play)(short buf[], int samples);
void (*play)(void *buf, int samples);
void (*stop)(void);
// may be null if not implemented
audio_alsa.c
+6 -4
View File
@@ -40,7 +40,7 @@ static void help(void);
static int init(int argc, char **argv);
static void deinit(void);
static void start(int i_sample_rate, int i_sample_format);
static void play(short buf[], int samples);
static void play(void *buf, int samples);
static void stop(void);
static void flush(void);
int delay(long *the_delay);
@@ -836,7 +836,7 @@ int delay(long *the_delay) {
}
}
static void play(short buf[], int samples) {
static void play(void *buf, int samples) {
// debug(3,"audio_alsa play called.");
int ret = 0;
if (alsa_handle == NULL) {
@@ -867,7 +867,7 @@ static void play(short buf[], int samples) {
if (samples == 0)
debug(1, "empty buffer being passed to pcm_writei -- skipping it");
if ((samples != 0) && (buf != NULL)) {
err = alsa_pcm_write(alsa_handle, (char *)buf, samples);
err = alsa_pcm_write(alsa_handle, buf, samples);
if (err < 0) {
debug(1, "Error %d writing %d samples in play(): \"%s\".", err, samples,
snd_strerror(err));
@@ -892,8 +892,9 @@ static void flush(void) {
int derr;
do_mute(1);
if (alsa_handle) {
if ((derr = snd_pcm_drop(alsa_handle)))
debug(1, "Error %d (\"%s\") draining the output device.", derr, snd_strerror(derr));
debug(1, "Error %d (\"%s\") dropping output device.", derr, snd_strerror(derr));
if ((derr = snd_pcm_hw_free(alsa_handle)))
debug(1, "Error %d (\"%s\") freeing the output device hardware.", derr, snd_strerror(derr));
@@ -901,6 +902,7 @@ static void flush(void) {
// flush also closes the device
if ((derr = snd_pcm_close(alsa_handle)))
debug(1, "Error %d (\"%s\") closing the output device.", derr, snd_strerror(derr));
alsa_handle = NULL;
}
pthread_mutex_unlock(&alsa_mutex);
player.c
+37 -24
View File
@@ -452,8 +452,7 @@ static void terminate_decoders(rtsp_conn_info *conn) {
static void init_buffer(rtsp_conn_info *conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
conn->audio_buffer[i].data = malloc(
conn->input_bytes_per_frame * (conn->max_frames_per_packet + conn->max_frame_size_change));
conn->audio_buffer[i].data = malloc(conn->input_bytes_per_frame * conn->max_frames_per_packet);
ab_resync(conn);
}
@@ -1036,13 +1035,13 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// ab_write),ab_read,ab_write);
conn->ab_buffering = 0;
}
signed short *silence;
void *silence;
// if (fs==0)
// debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld and
// dac_delay of %lld.",gross_frame_gap,dac_delay);
// the fs (number of frames of silence to play) can be zero in the DAC doesn't
// start
// ouotputting frames for a while -- it could get loaded up but not start
// outputting frames for a while -- it could get loaded up but not start
// responding
// for many milliseconds.
if (fs > 0) {
@@ -1066,7 +1065,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// debug(1,"Back end has no delay function.");
// send the appropriate prefiller here...
signed short *silence;
void *silence;
if (lead_time != 0) {
int64_t frame_gap = (lead_time * config.output_rate) >> 32;
// debug(1,"%d frames needed.",frame_gap);
@@ -1079,7 +1078,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (silence == NULL)
debug(1, "Failed to allocate %d frame silence buffer.", fs);
else {
// debug(1, "Outputting %d frames of silence.", fs);
// debug(1, "No delay function -- outputting %d frames of silence.", fs);
memset(silence, 0, conn->output_bytes_per_frame * fs);
config.output->play(silence, fs);
free(silence);
@@ -1546,7 +1545,7 @@ static void *player_thread_func(void *arg) {
static char rnstate[256];
initstate(time(NULL), rnstate, 256);
signed short *inbuf, *tbuf, *silence;
signed short *inbuf, *tbuf;
int32_t *sbuf;
@@ -1597,13 +1596,12 @@ static void *player_thread_func(void *arg) {
die("Failed to allocate memory for the transition buffer.");
sbuf = 0;
// initialise this, because soxr stuffing might be chosen later
// if (config.packet_stuffing == ST_soxr) { // needed for stuffing
sbuf = malloc(sizeof(int32_t) * 2 * (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 sbuf buffer.");
// }
// 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(
@@ -1611,12 +1609,6 @@ static void *player_thread_func(void *arg) {
(conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
if (outbuf == NULL)
die("Failed to allocate memory for an output buffer.");
silence = malloc(conn->output_bytes_per_frame * conn->max_frames_per_packet *
conn->output_sample_ratio);
if (silence == NULL)
die("Failed to allocate memory for a silence buffer.");
memset(silence, 0,
conn->output_bytes_per_frame * conn->max_frames_per_packet * conn->output_sample_ratio);
conn->first_packet_timestamp = 0;
conn->missing_packets = conn->late_packets = conn->too_late_packets = conn->resend_requests = 0;
conn->flush_rtp_timestamp =
@@ -1695,10 +1687,22 @@ static void *player_thread_func(void *arg) {
conn->play_number_after_flush++;
if (inframe->timestamp == 0) {
debug(3, "Player has supplied a silent frame, (possibly frame %u) for play number %d.",
SUCCESSOR(conn->last_seqno_read),play_number);
SUCCESSOR(conn->last_seqno_read), play_number);
conn->last_seqno_read = (SUCCESSOR(conn->last_seqno_read) &
0xffff); // manage the packet out of sequence minder
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
void *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 silent frame silence buffer.");
} else {
// the player may change the contents of the buffer, so it has to be zeroed each time;
// might as well malloc and freee it locally
memset(silence, 0, conn->output_bytes_per_frame * conn->max_frames_per_packet *
conn->output_sample_ratio);
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
free(silence);
}
} else if (conn->play_number_after_flush < 10) {
/*
int64_t difference = 0;
@@ -1708,7 +1712,18 @@ static void *player_thread_func(void *arg) {
debug(1, "Play number %d, monotonic timestamp %llx, difference
%lld.",conn->play_number_after_flush,inframe->timestamp,difference);
*/
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
void *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 flush silence buffer.");
} else {
// the player may change the contents of the buffer, so it has to be zeroed each time;
// might as well malloc and freee it locally
memset(silence, 0, conn->output_bytes_per_frame * conn->max_frames_per_packet *
conn->output_sample_ratio);
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
free(silence);
}
} else if (frames_to_drop) {
if (frames_to_drop > 3 * config.output_rate)
warn("Very large number of frames to drop: %" PRId64 ".", frames_to_drop);
@@ -1948,7 +1963,7 @@ static void *player_thread_func(void *arg) {
die("Failed to allocate memory for a long_silence buffer of %d frames.",
silence_length);
memset(long_silence, 0, conn->output_bytes_per_frame * silence_length);
config.output->play((short *)long_silence, silence_length);
config.output->play(long_silence, silence_length);
free(long_silence);
}
} else {
@@ -2085,7 +2100,7 @@ static void *player_thread_func(void *arg) {
if (play_samples == 0)
debug(1, "play_samples==0 skipping it (1).");
else
config.output->play((short *)outbuf, play_samples); // remove the (short*)!
config.output->play(outbuf, play_samples);
}
// check for loss of sync
@@ -2119,7 +2134,7 @@ static void *player_thread_func(void *arg) {
if (outbuf == NULL)
debug(1, "NULL outbuf to play -- skipping it.");
else
config.output->play((short *)outbuf, play_samples); // remove the (short*)!
config.output->play(outbuf, play_samples); // remove the (short*)!
}
// mark the frame as finished
@@ -2314,8 +2329,6 @@ static void *player_thread_func(void *arg) {
}
if (outbuf)
free(outbuf);
if (silence)
free(silence);
if (tbuf)
free(tbuf);
if (sbuf)
rtp.c
+7 -5
View File
@@ -302,13 +302,15 @@ void *rtp_control_receiver(void *arg) {
uint16_t flags = nctohs(&packet[2]);
int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency;
debug(3, "Latency derived just from the sync packet is %" PRId64 " frames.", la);
// debug(3, "Latency derived just from the sync packet is %" PRId64 " frames.", la);
if ((flags == 7) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion <= 353))) {
la += config.fixedLatencyOffset;
debug(3, "A fixed latency offset of %d frames has been added, giving a latency of "
"%" PRId64
" frames with flags: %d and AirPlay version %d (triggers if 353 or less).",
config.fixedLatencyOffset, la, flags, conn->AirPlayVersion);
// debug(3, "A fixed latency offset of %d frames has been added, giving a latency of
// "
// "%" PRId64
// " frames with flags: %d and AirPlay version %d (triggers if 353 or
// less).",
// config.fixedLatencyOffset, la, flags, conn->AirPlayVersion);
}
if ((conn->maximum_latency) && (conn->maximum_latency < la))
la = conn->maximum_latency;
rtsp.c
+3 -3
View File
@@ -1593,15 +1593,15 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag
if (hdr) {
debug(2, "Play connection from user agent \"%s\" on RTSP conversation thread %d.", hdr,
conn->connection_number);
// if the user agent is AirPlay and has a version number of 353 or less (from iOS 11.1)
// if the user agent is AirPlay and has a version number of 353 or less (from iOS 11.1,2)
// use the older way of calculating the latency
char *p = strstr(hdr, "AirPlay");
if (p) {
p = strchr(p, '/');
if (p) {
conn->AirPlayVersion = atoi(p + 1); // unsigned integer -- up to 2^32-1
debug(1, "AirPlay version %d detected.", conn->AirPlayVersion);
conn->AirPlayVersion = atoi(p + 1);
debug(2, "AirPlay version %d detected.", conn->AirPlayVersion);
}
}