diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c index 881d6f0af2..2dbfeca020 100644 --- a/fftools/ffmpeg.c +++ b/fftools/ffmpeg.c @@ -2017,11 +2017,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output, if (ret < 0) *decode_failed = 1; - if (ret >= 0 && avctx->sample_rate <= 0) { - av_log(avctx, AV_LOG_ERROR, "Sample rate %d invalid\n", avctx->sample_rate); - ret = AVERROR_INVALIDDATA; - } - if (ret != AVERROR_EOF) check_decode_result(ist, got_output, ret); @@ -2034,9 +2029,9 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output, /* increment next_dts to use for the case where the input stream does not have timestamps or there are multiple frames in the packet */ ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) / - avctx->sample_rate; + decoded_frame->sample_rate; ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) / - avctx->sample_rate; + decoded_frame->sample_rate; if (decoded_frame->pts != AV_NOPTS_VALUE) { decoded_frame_tb = ist->st->time_base; @@ -2054,8 +2049,10 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output, ist->prev_pkt_pts = pkt->pts; if (decoded_frame->pts != AV_NOPTS_VALUE) decoded_frame->pts = av_rescale_delta(decoded_frame_tb, decoded_frame->pts, - (AVRational){1, avctx->sample_rate}, decoded_frame->nb_samples, &ist->filter_in_rescale_delta_last, - (AVRational){1, avctx->sample_rate}); + (AVRational){1, decoded_frame->sample_rate}, + decoded_frame->nb_samples, + &ist->filter_in_rescale_delta_last, + (AVRational){1, decoded_frame->sample_rate}); ist->nb_samples = decoded_frame->nb_samples; err = send_frame_to_filters(ist, decoded_frame);