mirror of
https://github.com/sle118/squeezelite-esp32.git
synced 2025-12-08 04:27:12 +03:00
maximize number of AirPlay RTP buffers
This commit is contained in:
@@ -71,7 +71,8 @@ static log_level *loglevel = &raop_loglevel;
|
||||
//#define __RTP_STORE
|
||||
|
||||
// default buffer size
|
||||
#define BUFFER_FRAMES ( (150 * RAOP_SAMPLE_RATE * 2) / (352 * 100) )
|
||||
#define BUFFER_FRAMES_MAX ((RAOP_SAMPLE_RATE * 10) / 352 )
|
||||
#define BUFFER_FRAMES_MIN ( (150 * RAOP_SAMPLE_RATE * 2) / (352 * 100) )
|
||||
#define MAX_PACKET 1408
|
||||
#define MIN_LATENCY 11025
|
||||
#define MAX_LATENCY ( (120 * RAOP_SAMPLE_RATE * 2) / 100 )
|
||||
@@ -86,14 +87,15 @@ static log_level *loglevel = &raop_loglevel;
|
||||
enum { DATA = 0, CONTROL, TIMING };
|
||||
|
||||
static const u8_t silence_frame[MAX_PACKET] = { 0 };
|
||||
uint32_t buffer_frames = ((150 * RAOP_SAMPLE_RATE * 2) / (352 * 100));
|
||||
|
||||
typedef u16_t seq_t;
|
||||
typedef struct audio_buffer_entry { // decoded audio packets
|
||||
int ready;
|
||||
typedef struct __attribute__((__packed__)) audio_buffer_entry { // decoded audio packets
|
||||
u32_t rtptime, last_resend;
|
||||
s16_t *data;
|
||||
int len;
|
||||
bool allocated;
|
||||
u16_t len;
|
||||
u8_t ready;
|
||||
u8_t allocated;
|
||||
} abuf_t;
|
||||
|
||||
typedef struct rtp_s {
|
||||
@@ -133,7 +135,7 @@ typedef struct rtp_s {
|
||||
u32_t resent_req, resent_rec; // total resent + recovered frames
|
||||
u32_t silent_frames; // total silence frames
|
||||
u32_t discarded;
|
||||
abuf_t audio_buffer[BUFFER_FRAMES];
|
||||
abuf_t audio_buffer[BUFFER_FRAMES_MAX];
|
||||
seq_t ab_read, ab_write;
|
||||
pthread_mutex_t ab_mutex;
|
||||
#ifdef WIN32
|
||||
@@ -152,7 +154,7 @@ typedef struct rtp_s {
|
||||
} rtp_t;
|
||||
|
||||
|
||||
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
|
||||
#define BUFIDX(seqno) ((seq_t)(seqno) % buffer_frames)
|
||||
static void buffer_alloc(abuf_t *audio_buffer, int size, uint8_t *buf, size_t buf_size);
|
||||
static void buffer_release(abuf_t *audio_buffer);
|
||||
static void buffer_reset(abuf_t *audio_buffer);
|
||||
@@ -373,25 +375,27 @@ void rtp_record(rtp_t *ctx, unsigned short seqno, unsigned rtptime) {
|
||||
|
||||
/*---------------------------------------------------------------------------*/
|
||||
static void buffer_alloc(abuf_t *audio_buffer, int size, uint8_t *buf, size_t buf_size) {
|
||||
int i;
|
||||
for (i = 0; i < BUFFER_FRAMES; i++) {
|
||||
if (buf && buf_size >= size) {
|
||||
audio_buffer[i].data = (s16_t*) buf;
|
||||
audio_buffer[i].allocated = false;
|
||||
buf += size;
|
||||
buf_size -= size;
|
||||
} else {
|
||||
audio_buffer[i].allocated = true;
|
||||
audio_buffer[i].data = malloc(size);
|
||||
}
|
||||
audio_buffer[i].ready = 0;
|
||||
for (buffer_frames = 0; buf && buf_size >= size && buffer_frames < BUFFER_FRAMES_MAX; buffer_frames++) {
|
||||
audio_buffer[buffer_frames].data = (s16_t*) buf;
|
||||
audio_buffer[buffer_frames].allocated = 0;
|
||||
audio_buffer[buffer_frames].ready = 0;
|
||||
buf += size;
|
||||
buf_size -= size;
|
||||
}
|
||||
|
||||
LOG_INFO("allocated %d buffers (min=%d) from buffer of %zu bytes", buffer_frames, BUFFER_FRAMES_MIN, buf_size + buffer_frames * size);
|
||||
|
||||
for(; buffer_frames < BUFFER_FRAMES_MIN; buffer_frames++) {
|
||||
audio_buffer[buffer_frames].data = malloc(size);
|
||||
audio_buffer[buffer_frames].allocated = 1;
|
||||
audio_buffer[buffer_frames].ready = 0;
|
||||
}
|
||||
}
|
||||
|
||||
/*---------------------------------------------------------------------------*/
|
||||
static void buffer_release(abuf_t *audio_buffer) {
|
||||
int i;
|
||||
for (i = 0; i < BUFFER_FRAMES; i++) {
|
||||
for (i = 0; i < buffer_frames; i++) {
|
||||
if (audio_buffer[i].allocated) free(audio_buffer[i].data);
|
||||
}
|
||||
}
|
||||
@@ -399,7 +403,7 @@ static void buffer_release(abuf_t *audio_buffer) {
|
||||
/*---------------------------------------------------------------------------*/
|
||||
static void buffer_reset(abuf_t *audio_buffer) {
|
||||
int i;
|
||||
for (i = 0; i < BUFFER_FRAMES; i++) audio_buffer[i].ready = 0;
|
||||
for (i = 0; i < buffer_frames; i++) audio_buffer[i].ready = 0;
|
||||
}
|
||||
|
||||
/*---------------------------------------------------------------------------*/
|
||||
@@ -411,7 +415,7 @@ static int seq_order(seq_t a, seq_t b) {
|
||||
}
|
||||
|
||||
/*---------------------------------------------------------------------------*/
|
||||
static void alac_decode(rtp_t *ctx, s16_t *dest, char *buf, int len, int *outsize) {
|
||||
static void alac_decode(rtp_t *ctx, s16_t *dest, char *buf, int len, u16_t *outsize) {
|
||||
unsigned char iv[16];
|
||||
int aeslen;
|
||||
assert(len<=MAX_PACKET);
|
||||
@@ -803,7 +807,7 @@ 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
|
||||
|
||||
// do not request silly ranges (happens in case of network large blackouts)
|
||||
// do not request silly ranges (happens in case of network large blackouts)
|
||||
if (seq_order(last, first) || last - first > buffer_frames / 2) return false;
|
||||
|
||||
ctx->resent_req += (seq_t) (last - first) + 1;
|
||||
|
||||
Reference in New Issue
Block a user