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https://github.com/FFmpeg/FFmpeg.git
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alacdec: support 32 bps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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parent
8d95eb6702
commit
89fc7e36c7
@ -351,6 +351,17 @@ static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
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}
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}
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static void interleave_stereo_32(int32_t *buffer[MAX_CHANNELS],
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int32_t *buffer_out, int numsamples)
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{
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int i;
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for (i = 0; i < numsamples; i++) {
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*buffer_out++ = buffer[0][i];
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*buffer_out++ = buffer[1][i];
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}
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}
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static int alac_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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@ -533,6 +544,16 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
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outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
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}
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break;
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case 32:
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if (channels == 2) {
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interleave_stereo_32(alac->outputsamples_buffer,
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(int32_t *)alac->frame.data[0], outputsamples);
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} else {
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int32_t *outbuffer = (int32_t *)alac->frame.data[0];
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for (i = 0; i < outputsamples; i++)
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outbuffer[i] = alac->outputsamples_buffer[0][i];
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}
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break;
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}
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if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
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@ -628,6 +649,7 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
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switch (alac->setinfo_sample_size) {
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case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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break;
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case 32:
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case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
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break;
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default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
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