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@@ -401,6 +401,8 @@ int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int a | @@ -401,6 +401,8 @@ int64_t SrsHlsAacJitter::on_buffer_start(int64_t flv_pts, int sample_rate, int a | ||
401 | } | 401 | } |
402 | 402 | ||
403 | // @see: ngx_rtmp_hls_audio | 403 | // @see: ngx_rtmp_hls_audio |
404 | + // drop the rtmp audio packet timestamp, re-calc it by sample rate. | ||
405 | + // | ||
404 | // resample for the tbn of ts is 90000, flv is 1000, | 406 | // resample for the tbn of ts is 90000, flv is 1000, |
405 | // we will lost timestamp if use audio packet timestamp, | 407 | // we will lost timestamp if use audio packet timestamp, |
406 | // so we must resample. or audio will corupt in IOS. | 408 | // so we must resample. or audio will corupt in IOS. |
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