X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;ds=sidebyside;f=ffplay.c;h=749cfc7b9127fb46e9ccfdf2a2e3fcffbf284d76;hb=73da4fd520379adafe858b4e182a3eac0f49080c;hp=bd7374a3ff81b951f27365c3c85a7e7102d003af;hpb=794006f8fe403ccd2993513a8e4fbe885ef7f9fb;p=ffmpeg diff --git a/ffplay.c b/ffplay.c index bd7374a3ff8..749cfc7b912 100644 --- a/ffplay.c +++ b/ffplay.c @@ -1970,25 +1970,19 @@ static void update_sample_display(VideoState *is, short *samples, int samples_si } } -/* return the new audio buffer size (samples can be added or deleted - to get better sync if video or external master clock) */ -static int synchronize_audio(VideoState *is, short *samples, - int samples_size1, double pts) +/* return the wanted number of samples to get better sync if sync_type is video + * or external master clock */ +static int synchronize_audio(VideoState *is, int nb_samples) { - int n, samples_size; - double ref_clock; - - n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels; - samples_size = samples_size1; + int wanted_nb_samples = nb_samples; /* if not master, then we try to remove or add samples to correct the clock */ if (((is->av_sync_type == AV_SYNC_VIDEO_MASTER && is->video_st) || is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) { double diff, avg_diff; - int wanted_size, min_size, max_size, nb_samples; + int min_nb_samples, max_nb_samples; - ref_clock = get_master_clock(is); - diff = get_audio_clock(is) - ref_clock; + diff = get_audio_clock(is) - get_master_clock(is); if (diff < AV_NOSYNC_THRESHOLD) { is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum; @@ -2000,38 +1994,13 @@ static int synchronize_audio(VideoState *is, short *samples, avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef); if (fabs(avg_diff) >= is->audio_diff_threshold) { - wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n); - nb_samples = samples_size / n; - - min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; - max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; - if (wanted_size < min_size) - wanted_size = min_size; - else if (wanted_size > FFMIN3(max_size, samples_size, sizeof(is->audio_buf2))) - wanted_size = FFMIN3(max_size, samples_size, sizeof(is->audio_buf2)); - - /* add or remove samples to correction the synchro */ - if (wanted_size < samples_size) { - /* remove samples */ - samples_size = wanted_size; - } else if (wanted_size > samples_size) { - uint8_t *samples_end, *q; - int nb; - - /* add samples */ - nb = (samples_size - wanted_size); - samples_end = (uint8_t *)samples + samples_size - n; - q = samples_end + n; - while (nb > 0) { - memcpy(q, samples_end, n); - q += n; - nb -= n; - } - samples_size = wanted_size; - } + wanted_nb_samples = nb_samples + (int)(diff * is->audio_src_freq); + min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100)); + max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100)); + wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples); } av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n", - diff, avg_diff, samples_size - samples_size1, + diff, avg_diff, wanted_nb_samples - nb_samples, is->audio_clock, is->video_clock, is->audio_diff_threshold); } } else { @@ -2042,7 +2011,7 @@ static int synchronize_audio(VideoState *is, short *samples, } } - return samples_size; + return wanted_nb_samples; } /* decode one audio frame and returns its uncompressed size */ @@ -2057,6 +2026,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) double pts; int new_packet = 0; int flush_complete = 0; + int wanted_nb_samples; for (;;) { /* NOTE: the audio packet can contain several frames */ @@ -2091,8 +2061,12 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) dec->sample_fmt, 1); dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels); + wanted_nb_samples = synchronize_audio(is, is->frame->nb_samples); - if (dec->sample_fmt != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || dec->sample_rate != is->audio_src_freq) { + if (dec->sample_fmt != is->audio_src_fmt || + dec_channel_layout != is->audio_src_channel_layout || + dec->sample_rate != is->audio_src_freq || + (wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) { if (is->swr_ctx) swr_free(&is->swr_ctx); is->swr_ctx = swr_alloc_set_opts(NULL, @@ -2119,8 +2093,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) if (is->swr_ctx) { const uint8_t *in[] = { is->frame->data[0] }; uint8_t *out[] = {is->audio_buf2}; + if (wanted_nb_samples != is->frame->nb_samples) { + if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt_freq / dec->sample_rate, + wanted_nb_samples * is->audio_tgt_freq / dec->sample_rate) < 0) { + fprintf(stderr, "swr_set_compensation() failed\n"); + break; + } + } len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt), - in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt)); + in, is->frame->nb_samples); if (len2 < 0) { fprintf(stderr, "audio_resample() failed\n"); break; @@ -2182,6 +2163,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) VideoState *is = opaque; int audio_size, len1; int bytes_per_sec; + int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt_channels, 1, is->audio_tgt_fmt, 1); double pts; audio_callback_time = av_gettime(); @@ -2192,12 +2174,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) if (audio_size < 0) { /* if error, just output silence */ is->audio_buf = is->silence_buf; - is->audio_buf_size = sizeof(is->silence_buf); + is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size; } else { if (is->show_mode != SHOW_MODE_VIDEO) update_sample_display(is, (int16_t *)is->audio_buf, audio_size); - audio_size = synchronize_audio(is, (int16_t *)is->audio_buf, audio_size, - pts); is->audio_buf_size = audio_size; } is->audio_buf_index = 0;