ffplay: use avctx->channels and avctx->freq before avcodec_open2 consistently
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parent
1e7f7dc2d1
commit
2446a8cc0a
17
ffplay.c
17
ffplay.c
@ -2183,8 +2183,15 @@ static int stream_component_open(VideoState *is, int stream_index)
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if(codec->capabilities & CODEC_CAP_DR1)
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avctx->flags |= CODEC_FLAG_EMU_EDGE;
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wanted_spec.freq = avctx->sample_rate;
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wanted_spec.channels = avctx->channels;
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if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
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if(avctx->sample_rate <= 0 || avctx->channels <= 0){
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fprintf(stderr, "Invalid sample rate or channel count\n");
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return -1;
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}
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wanted_spec.freq = avctx->sample_rate;
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wanted_spec.channels = avctx->channels;
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}
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if (!codec ||
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avcodec_open2(avctx, codec, &opts) < 0)
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return -1;
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@ -2195,10 +2202,6 @@ static int stream_component_open(VideoState *is, int stream_index)
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/* prepare audio output */
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if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
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if(avctx->sample_rate <= 0 || avctx->channels <= 0){
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fprintf(stderr, "Invalid sample rate or channel count\n");
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return -1;
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}
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wanted_spec.format = AUDIO_S16SYS;
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wanted_spec.silence = 0;
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wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
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@ -2225,7 +2228,7 @@ static int stream_component_open(VideoState *is, int stream_index)
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is->audio_diff_avg_count = 0;
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/* since we do not have a precise anough audio fifo fullness,
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we correct audio sync only if larger than this threshold */
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is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate;
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is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / wanted_spec.freq;
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memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
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packet_queue_init(&is->audioq);
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