diff --git a/ffplay.c b/ffplay.c index eee9da3576..2aaef3590b 100644 --- a/ffplay.c +++ b/ffplay.c @@ -1828,9 +1828,10 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for return AVERROR(ENOMEM); ret = snprintf(asrc_args, sizeof(asrc_args), - "sample_rate=%d:sample_fmt=%s:channels=%d", + "sample_rate=%d:sample_fmt=%s:channels=%d:time_base=%d/%d", is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt), - is->audio_filter_src.channels); + is->audio_filter_src.channels, + 1, is->audio_filter_src.freq); if (is->audio_filter_src.channel_layout) snprintf(asrc_args + ret, sizeof(asrc_args) - ret, ":channel_layout=0x%"PRIx64, is->audio_filter_src.channel_layout); @@ -2190,11 +2191,13 @@ static int audio_decode_frame(VideoState *is) continue; } + tb = (AVRational){1, is->frame->sample_rate}; + if (is->frame->pts != AV_NOPTS_VALUE) + is->frame->pts = av_rescale_q(is->frame->pts, dec->time_base, tb); if (is->frame->pts == AV_NOPTS_VALUE && pkt_temp->pts != AV_NOPTS_VALUE) - is->frame->pts = av_rescale_q(pkt_temp->pts, is->audio_st->time_base, dec->time_base); + is->frame->pts = av_rescale_q(pkt_temp->pts, is->audio_st->time_base, tb); if (pkt_temp->pts != AV_NOPTS_VALUE) pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base); - tb = dec->time_base; #if CONFIG_AVFILTER {