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Cosmetics: indentation
Originally committed as revision 9729 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -492,43 +492,43 @@ static int alac_decode_frame(AVCodecContext *avctx,
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channels = get_bits(&alac->gb, 3) + 1;
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/* 2^result = something to do with output waiting.
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* perhaps matters if we read > 1 frame in a pass?
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*/
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get_bits(&alac->gb, 4);
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/* 2^result = something to do with output waiting.
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* perhaps matters if we read > 1 frame in a pass?
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*/
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get_bits(&alac->gb, 4);
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get_bits(&alac->gb, 12); /* unknown, skip 12 bits */
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get_bits(&alac->gb, 12); /* unknown, skip 12 bits */
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hassize = get_bits(&alac->gb, 1); /* the output sample size is stored soon */
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/* the output sample size is stored soon */
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hassize = get_bits(&alac->gb, 1);
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wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
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wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
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isnotcompressed = get_bits(&alac->gb, 1); /* whether the frame is compressed */
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/* whether the frame is compressed */
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isnotcompressed = get_bits(&alac->gb, 1);
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if (hassize) {
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/* now read the number of samples,
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* as a 32bit integer */
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outputsamples = get_bits(&alac->gb, 32);
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} else
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outputsamples = alac->setinfo_max_samples_per_frame;
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if (hassize) {
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/* now read the number of samples as a 32bit integer */
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outputsamples = get_bits(&alac->gb, 32);
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} else
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outputsamples = alac->setinfo_max_samples_per_frame;
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*outputsize = outputsamples * alac->bytespersample;
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readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
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*outputsize = outputsamples * alac->bytespersample;
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readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
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if (!isnotcompressed) {
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/* so it is compressed */
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int16_t predictor_coef_table[channels][32];
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int predictor_coef_num[channels];
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int prediction_type[channels];
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int prediction_quantitization[channels];
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int ricemodifier[channels];
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if (!isnotcompressed) {
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/* so it is compressed */
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int16_t predictor_coef_table[channels][32];
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int predictor_coef_num[channels];
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int prediction_type[channels];
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int prediction_quantitization[channels];
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int ricemodifier[channels];
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int i, chan;
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int i, chan;
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interlacing_shift = get_bits(&alac->gb, 8);
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interlacing_leftweight = get_bits(&alac->gb, 8);
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interlacing_shift = get_bits(&alac->gb, 8);
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interlacing_leftweight = get_bits(&alac->gb, 8);
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for (chan = 0; chan < channels; chan++) {
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for (chan = 0; chan < channels; chan++) {
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prediction_type[chan] = get_bits(&alac->gb, 4);
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prediction_quantitization[chan] = get_bits(&alac->gb, 4);
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@ -539,13 +539,13 @@ static int alac_decode_frame(AVCodecContext *avctx,
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for (i = 0; i < predictor_coef_num[chan]; i++) {
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predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
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}
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}
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}
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if (wasted_bytes) {
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av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
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}
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if (wasted_bytes) {
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av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
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}
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for (chan = 0; chan < channels; chan++) {
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for (chan = 0; chan < channels; chan++) {
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bastardized_rice_decompress(alac,
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alac->predicterror_buffer[chan],
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outputsamples,
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@ -556,7 +556,7 @@ static int alac_decode_frame(AVCodecContext *avctx,
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(1 << alac->setinfo_rice_kmodifier) - 1);
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if (prediction_type[chan] == 0) {
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/* adaptive fir */
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/* adaptive fir */
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predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
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alac->outputsamples_buffer[chan],
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outputsamples,
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@ -573,12 +573,12 @@ static int alac_decode_frame(AVCodecContext *avctx,
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* little strange..
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*/
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}
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}
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} else {
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/* not compressed, easy case */
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if (alac->setinfo_sample_size <= 16) {
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int i, chan;
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for (chan = 0; chan < channels; chan++) {
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}
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} else {
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/* not compressed, easy case */
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if (alac->setinfo_sample_size <= 16) {
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int i, chan;
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for (chan = 0; chan < channels; chan++) {
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for (i = 0; i < outputsamples; i++) {
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int32_t audiobits;
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@ -587,10 +587,10 @@ static int alac_decode_frame(AVCodecContext *avctx,
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alac->outputsamples_buffer[chan][i] = audiobits;
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}
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}
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} else {
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int i, chan;
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for (chan = 0; chan < channels; chan++) {
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}
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} else {
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int i, chan;
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for (chan = 0; chan < channels; chan++) {
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for (i = 0; i < outputsamples; i++) {
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int32_t audiobits;
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@ -603,16 +603,16 @@ static int alac_decode_frame(AVCodecContext *avctx,
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alac->outputsamples_buffer[chan][i] = audiobits;
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}
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}
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}
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/* wasted_bytes = 0; */
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interlacing_shift = 0;
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interlacing_leftweight = 0;
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}
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/* wasted_bytes = 0; */
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interlacing_shift = 0;
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interlacing_leftweight = 0;
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}
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switch(alac->setinfo_sample_size) {
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case 16: {
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if (channels == 2) {
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switch(alac->setinfo_sample_size) {
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case 16: {
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if (channels == 2) {
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deinterlace_16(alac->outputsamples_buffer[0],
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alac->outputsamples_buffer[1],
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(int16_t*)outbuffer,
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@ -620,24 +620,23 @@ static int alac_decode_frame(AVCodecContext *avctx,
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outputsamples,
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interlacing_shift,
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interlacing_leftweight);
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} else {
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int i;
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for (i = 0; i < outputsamples; i++) {
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int16_t sample = alac->outputsamples_buffer[0][i];
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((int16_t*)outbuffer)[i * alac->numchannels] = sample;
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}
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}
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break;
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} else {
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int i;
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for (i = 0; i < outputsamples; i++) {
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int16_t sample = alac->outputsamples_buffer[0][i];
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((int16_t*)outbuffer)[i * alac->numchannels] = sample;
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}
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}
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case 20:
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case 24:
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case 32:
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av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
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break;
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default:
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break;
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}
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break;
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}
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case 20:
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case 24:
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case 32:
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av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
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break;
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default:
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break;
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}
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return input_buffer_size;
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}
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