ffplay: use avctx->channels and avctx->freq before avcodec_open2 consistently

This commit is contained in:
Marton Balint 2011-10-02 11:00:40 +02:00
parent 1e7f7dc2d1
commit 2446a8cc0a

View File

@ -2183,8 +2183,15 @@ static int stream_component_open(VideoState *is, int stream_index)
if(codec->capabilities & CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
wanted_spec.freq = avctx->sample_rate;
wanted_spec.channels = avctx->channels;
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if(avctx->sample_rate <= 0 || avctx->channels <= 0){
fprintf(stderr, "Invalid sample rate or channel count\n");
return -1;
}
wanted_spec.freq = avctx->sample_rate;
wanted_spec.channels = avctx->channels;
}
if (!codec ||
avcodec_open2(avctx, codec, &opts) < 0)
return -1;
@ -2195,10 +2202,6 @@ static int stream_component_open(VideoState *is, int stream_index)
/* prepare audio output */
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if(avctx->sample_rate <= 0 || avctx->channels <= 0){
fprintf(stderr, "Invalid sample rate or channel count\n");
return -1;
}
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
@ -2225,7 +2228,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio fifo fullness,
we correct audio sync only if larger than this threshold */
is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate;
is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / wanted_spec.freq;
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
packet_queue_init(&is->audioq);