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https://github.com/sle118/squeezelite-esp32.git
synced 2025-12-06 03:27:01 +03:00
see CHANGELOG
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@@ -1,8 +1,13 @@
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2024-01-19
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- fixed libflac with OggFlac
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- AirPlay missed frame logging
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2024-01-16
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- catch-up with cspot latest
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- refactor airplay flush/first packet
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- new libFLAC that supports multi-stream OggFlac
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- fix output threshold
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- log missed frames
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2024-01-10
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- add OggFlac to stream metadata
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Binary file not shown.
@@ -96,6 +96,7 @@ typedef struct __attribute__((__packed__)) audio_buffer_entry { // decoded aud
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u16_t len;
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u8_t ready;
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u8_t allocated;
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u8_t missed;
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} abuf_t;
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typedef struct rtp_s {
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@@ -477,10 +478,11 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
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u32_t playtime = ctx->synchro.time + ((rtptime - ctx->synchro.rtp) * 10) / (RAOP_SAMPLE_RATE / 100);
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ctx->cmd_cb(RAOP_PLAY, playtime);
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}
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abuf = ctx->audio_buffer + BUFIDX(seqno);
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if (seqno == (u16_t) (ctx->ab_write+1)) {
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// expected packet
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abuf = ctx->audio_buffer + BUFIDX(seqno);
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ctx->ab_write = seqno;
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LOG_SDEBUG("packet expected seqno:%hu rtptime:%u (W:%hu R:%hu)", seqno, rtptime, ctx->ab_write, ctx->ab_read);
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} else if (seq_order(ctx->ab_write, seqno)) {
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@@ -504,16 +506,15 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
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LOG_DEBUG("[%p]: packet newer seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
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}
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abuf = ctx->audio_buffer + BUFIDX(seqno);
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ctx->ab_write = seqno;
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} else if (seq_order(ctx->ab_read, seqno + 1)) {
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// recovered packet, not yet sent
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abuf = ctx->audio_buffer + BUFIDX(seqno);
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ctx->resent_rec++;
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LOG_DEBUG("[%p]: packet recovered seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
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} else {
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// too late
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LOG_DEBUG("[%p]: packet too late seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
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// too late
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if (abuf->missed) LOG_INFO("[%p]: packet too late seqno:%hu rtptime:%u (W:%hu R:%hu)", ctx, seqno, rtptime, ctx->ab_write, ctx->ab_read);
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abuf = NULL;
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}
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if (ctx->in_frames++ > 1000) {
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@@ -524,6 +525,7 @@ static void buffer_put_packet(rtp_t *ctx, seq_t seqno, unsigned rtptime, bool fi
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if (abuf) {
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alac_decode(ctx, abuf->data, data, len, &abuf->len);
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abuf->ready = 1;
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abuf->missed = 0;
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// this is the local rtptime when this frame is expected to play
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abuf->rtptime = rtptime;
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buffer_push_packet(ctx);
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@@ -567,6 +569,7 @@ static void buffer_push_packet(rtp_t *ctx) {
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LOG_DEBUG("[%p]: created zero frame (W:%hu R:%hu)", ctx, ctx->ab_write, ctx->ab_read);
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ctx->data_cb(silence_frame, ctx->frame_size * 4, playtime);
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ctx->silent_frames++;
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curframe->missed = 1;
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}
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} else if (curframe->ready) {
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ctx->data_cb((const u8_t*) curframe->data, curframe->len, playtime);
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