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@ -107,6 +107,14 @@ static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize) |
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return HEADER_SIZE; |
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} |
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#if FF_API_OLD_ENCODE_AUDIO |
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static av_cold int adx_encode_close(AVCodecContext *avctx) |
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{ |
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av_freep(&avctx->coded_frame); |
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return 0; |
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} |
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#endif |
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static av_cold int adx_encode_init(AVCodecContext *avctx) |
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{ |
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ADXContext *c = avctx->priv_data; |
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@ -118,8 +126,8 @@ static av_cold int adx_encode_init(AVCodecContext *avctx) |
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avctx->frame_size = BLOCK_SAMPLES; |
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#if FF_API_OLD_ENCODE_AUDIO |
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avcodec_get_frame_defaults(&c->frame); |
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avctx->coded_frame = &c->frame; |
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if (!(avctx->coded_frame = avcodec_alloc_frame())) |
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return AVERROR(ENOMEM); |
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#endif |
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/* the cutoff can be adjusted, but this seems to work pretty well */ |
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@ -169,6 +177,9 @@ AVCodec ff_adpcm_adx_encoder = { |
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.id = AV_CODEC_ID_ADPCM_ADX, |
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.priv_data_size = sizeof(ADXContext), |
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.init = adx_encode_init, |
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#if FF_API_OLD_ENCODE_AUDIO |
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.close = adx_encode_close, |
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#endif |
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.encode2 = adx_encode_frame, |
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, |
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AV_SAMPLE_FMT_NONE }, |
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