From d0707677fa50f7d75a945e234287a4661962c851 Mon Sep 17 00:00:00 2001 From: Michael Niedermayer Date: Sat, 13 Oct 2012 16:45:01 +0200 Subject: ffplay: use audio parameters from the decoded frame instead of AVCodecContext Based on commit by Justin Ruggles (the changed code is too different to apply as is) Signed-off-by: Michael Niedermayer --- ffplay.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'ffplay.c') diff --git a/ffplay.c b/ffplay.c index 438c11d17b..5a2c7728fc 100644 --- a/ffplay.c +++ b/ffplay.c @@ -1974,34 +1974,34 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) flush_complete = 1; continue; } - data_size = av_samples_get_buffer_size(NULL, dec->channels, + data_size = av_samples_get_buffer_size(NULL, is->frame->channels, is->frame->nb_samples, - dec->sample_fmt, 1); + is->frame->format, 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); + (is->frame->channel_layout && is->frame->channels == av_get_channel_layout_nb_channels(is->frame->channel_layout)) ? + is->frame->channel_layout : av_get_default_channel_layout(is->frame->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 || - (wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) { + if (is->frame->format != is->audio_src.fmt || + dec_channel_layout != is->audio_src.channel_layout || + is->frame->sample_rate != is->audio_src.freq || + (wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) { swr_free(&is->swr_ctx); is->swr_ctx = swr_alloc_set_opts(NULL, is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq, - dec_channel_layout, dec->sample_fmt, dec->sample_rate, + dec_channel_layout, is->frame->format, is->frame->sample_rate, 0, NULL); if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) { fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n", - dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels, + is->frame->sample_rate, av_get_sample_fmt_name(is->frame->format), (int)is->frame->channels, is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels); break; } is->audio_src.channel_layout = dec_channel_layout; - is->audio_src.channels = dec->channels; - is->audio_src.freq = dec->sample_rate; - is->audio_src.fmt = dec->sample_fmt; + is->audio_src.channels = is->frame->channels; + is->audio_src.freq = is->frame->sample_rate; + is->audio_src.fmt = is->frame->format; } if (is->swr_ctx) { @@ -2009,8 +2009,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) uint8_t *out[] = {is->audio_buf2}; int out_count = sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt); 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) { + if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / is->frame->sample_rate, + wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate) < 0) { fprintf(stderr, "swr_set_compensation() failed\n"); break; } @@ -2035,7 +2035,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) pts = is->audio_clock; *pts_ptr = pts; is->audio_clock += (double)data_size / - (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt)); + (is->frame->channels * is->frame->sample_rate * av_get_bytes_per_sample(is->frame->format)); #ifdef DEBUG { static double last_clock; -- cgit v1.2.3