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airplay buffer management correction

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This commit is contained in:
philippe44
2019-08-25 23:05:21 -07:00
parent 8a6295a016
commit df525bbb10
3 changed files with 40 additions and 15 deletions
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48
components/raop/rtp.c
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@@ -211,9 +211,7 @@ rtp_resp_t rtp_init(struct in_addr host, int latency, char *aeskey, char *aesiv,
ctx->rtp_host.sin_family = AF_INET;
ctx->rtp_host.sin_addr.s_addr = INADDR_ANY;
pthread_mutex_init(&ctx->ab_mutex, 0);
ctx->flush_seqno = -1;
ctx->latency = latency;
ctx->flush_seqno = -1;
ctx->latency = latency;
ctx->ab_read = ctx->ab_write;
@@ -375,7 +373,7 @@ static void alac_decode(rtp_t *ctx, s16_t *dest, char *buf, int len, int *outsiz
/*---------------------------------------------------------------------------*/
static void alac_decode(rtp_t *ctx, s16_t *dest, char *buf, int len, int *outsize) {
unsigned char iv[16];
int aeslen;
int aeslen;
assert(len<=MAX_PACKET);
if (ctx->decrypt) {
@@ -413,7 +411,7 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
pthread_mutex_unlock(&ctx->ab_mutex);
return;
}
}
}
if (seqno == (u16_t) (ctx->ab_write+1)) {
// expected packet
@@ -421,20 +419,20 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
ctx->ab_write = seqno;
LOG_SDEBUG("packet expected seqno:%hu rtptime:%u (W:%hu R:%hu)", seqno, rtptime, ctx->ab_write, ctx->ab_read);
} else if (seq_order(ctx->ab_write, seqno)) {
} else if (seq_order(ctx->ab_write, seqno)) {
// newer than expected
if (seqno - ctx->ab_write - 1 > ctx->latency / ctx->frame_size) {
if (ctx->latency && seq_order(ctx->latency / ctx->frame_size, seqno - ctx->ab_write - 1)) {
// only get rtp latency-1 frames back (last one is seqno)
LOG_WARN("[%p] too many missing frames %hu seq: %hu, (W:%hu R:%hu)", ctx, seqno - ctx->ab_write - 1, seqno, ctx->ab_write, ctx->ab_read);
ctx->ab_write = seqno - ctx->latency / ctx->frame_size;
ctx->ab_write = seqno - ctx->latency / ctx->frame_size;
}
if (seqno - ctx->ab_read + 1 > ctx->latency / ctx->frame_size) {
if (ctx->latency && seq_order(ctx->latency / ctx->frame_size, seqno - ctx->ab_read)) {
// if ab_read is lagging more than http latency, advance it
LOG_WARN("[%p] on hold for too long %hu (W:%hu R:%hu)", ctx, seqno - ctx->ab_read + 1, ctx->ab_write, ctx->ab_read);
ctx->ab_read = seqno - ctx->latency / ctx->frame_size + 1;
}
if (rtp_request_resend(ctx, ctx->ab_write + 1, seqno-1)) {
seq_t i;
seq_t i;
u32_t now = gettime_ms();
for (i = ctx->ab_write + 1; seq_order(i, seqno); i++) {
ctx->audio_buffer[BUFIDX(i)].rtptime = rtptime - (seqno-i)*ctx->frame_size;
@@ -453,7 +451,7 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
// too late
LOG_DEBUG("[%p]: packet too late seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
}
if (ctx->in_frames++ > 1000) {
LOG_INFO("[%p]: fill [level:%hu rec:%u] [W:%hu R:%hu]", ctx, ctx->ab_write - ctx->ab_read, ctx->resent_rec, ctx->ab_write, ctx->ab_read);
ctx->in_frames = 0;
@@ -496,6 +494,27 @@ static void buffer_push_packet(rtp_t *ctx) {
if (now > playtime) {
LOG_DEBUG("[%p]: discarded frame now:%u missed by:%d (W:%hu R:%hu)", ctx, now, now - playtime, ctx->ab_write, ctx->ab_read);
ctx->discarded++;
} else if (curframe->ready) {
/*
// some dirty code to see if the click problem comes from i2s stage or decoder stage
static s16_t sin_data[200];
static bool gen = false;
if (!gen) {
for (i = 0; i < 200; i++) sin_data[i] = 1024 * sin((2*3.14159*220.5*i)/44100.);
gen = true;
}
static int c = 0;
int cnt = 0;
s16_t *p = (s16_t*) curframe->data;
while (cnt++ < 352) {
*p = sin_data[c++ % 200];
*(p+1) = *p;
p += 2;
}
curframe->len = 1408;
*/
ctx->data_cb((const u8_t*) curframe->data, curframe->len, playtime);
@@ -507,8 +526,7 @@ static void buffer_push_packet(rtp_t *ctx) {
} else break;
ctx->ab_read++;
ctx->out_frames++;
ctx->out_frames++;
} while (seq_order(ctx->ab_read, ctx->ab_write));
@@ -520,7 +538,7 @@ static void buffer_push_packet(rtp_t *ctx) {
}
LOG_SDEBUG("playtime %u %d [W:%hu R:%hu] %d", playtime, playtime - now, ctx->ab_write, ctx->ab_read, curframe->ready);
// each missing packet will be requested up to (latency_frames / 16) times
for (i = 0; seq_order(ctx->ab_read + i, ctx->ab_write); i += 16) {
abuf_t *frame = ctx->audio_buffer + BUFIDX(ctx->ab_read + i);
@@ -732,7 +750,7 @@ static bool rtp_request_resend(rtp_t *ctx, seq_t first, seq_t last) {
/*---------------------------------------------------------------------------*/
static bool rtp_request_resend(rtp_t *ctx, seq_t first, seq_t last) {
unsigned char req[8]; // *not* a standard RTCP NACK
unsigned char req[8]; // *not* a standard RTCP NACK
// do not request silly ranges (happens in case of network large blackouts)
if (seq_order(last, first) || last - first > BUFFER_FRAMES / 2) return false;