mirror of https://git.ffmpeg.org/ffmpeg.git
Support decoding of 24 bit True Audio samples-
Many samples will fail because "Output buffer size is too small."
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parent
4cf2e30f05
commit
7d70d1937a
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@ -265,7 +265,7 @@ static av_cold int tta_decode_init(AVCodecContext * avctx)
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else switch(s->bps) {
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case 1: avctx->sample_fmt = AV_SAMPLE_FMT_U8; break;
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case 2: avctx->sample_fmt = AV_SAMPLE_FMT_S16; break;
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// case 3: avctx->sample_fmt = AV_SAMPLE_FMT_S24; break;
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case 3: avctx->bits_per_coded_sample = 24;
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case 4: avctx->sample_fmt = AV_SAMPLE_FMT_S32; break;
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default:
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av_log_ask_for_sample(s->avctx,
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@ -459,6 +459,13 @@ static int tta_decode_frame(AVCodecContext *avctx,
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*data_size = (uint8_t *)samples - (uint8_t *)data;
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break;
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}
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case 3: {
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int32_t *samples = data;
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for (p = s->decode_buffer; p < s->decode_buffer + (framelen * s->channels); p++)
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*samples++ = AV_RN32(p) << 8;
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*data_size = (uint8_t *)samples - (uint8_t *)data;
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break;
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}
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default:
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av_log(s->avctx, AV_LOG_ERROR, "Error, only 16bit samples supported!\n");
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}
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