[FFmpeg-devel] [PATCH] ALAC Encoder
Michael Niedermayer
michaelni
Mon Aug 18 00:16:23 CEST 2008
On Mon, Aug 18, 2008 at 02:38:24AM +0530, Jai Menon wrote:
> Hi,
>
> On Sunday 17 Aug 2008 5:17:52 pm Michael Niedermayer wrote:
> > On Sun, Aug 17, 2008 at 11:17:10AM +0530, Jai Menon wrote:
[...]
[...]
> Index: libavcodec/alacenc.c
> ===================================================================
> --- libavcodec/alacenc.c (revision 14818)
> +++ libavcodec/alacenc.c (working copy)
> @@ -33,15 +33,58 @@
>
> #define ALAC_ESCAPE_CODE 0x1FF
> #define ALAC_MAX_LPC_ORDER 30
> +#define DEFAULT_MAX_PRED_ORDER 6
> +#define DEFAULT_MIN_PRED_ORDER 4
> +#define ALAC_MAX_LPC_PRECISION 9
> +#define ALAC_MAX_LPC_SHIFT 9
ok
>
> +#define ALAC_CHMODE_LEFT_RIGHT 1
> +#define ALAC_CHMODE_LEFT_SIDE 8
> +#define ALAC_CHMODE_RIGHT_SIDE 9
> +#define ALAC_CHMODE_MID_SIDE 10
> +
> +typedef struct RiceContext {
> + int history_mult;
> + int initial_history;
> + int k_modifier;
> + int rice_modifier;
> +} RiceContext;
> +
> +typedef struct LPCContext {
> + int lpc_order;
> + int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
> + int lpc_quant;
> +} LPCContext;
> +
> +typedef struct AlacEncodeContext {
> + int compression_level;
> + int max_coded_frame_size;
> + int write_sample_size;
> + int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
ok
> + int32_t predictor_buf[DEFAULT_FRAME_SIZE];
> int interlacing_shift;
> int interlacing_leftweight;
> PutBitContext pbctx;
> + RiceContext rc;
> + LPCContext lpc[MAX_CHANNELS];
ok
> DSPContext dspctx;
> AVCodecContext *avctx;
> } AlacEncodeContext;
>
>
> +static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
> +{
> + int ch, i;
> +
> + for(ch=0;ch<s->avctx->channels;ch++) {
> + int16_t *sptr = input_samples + ch;
> + for(i=0;i<s->avctx->frame_size;i++) {
> + s->sample_buf[ch][i] = *sptr;
> + sptr += s->avctx->channels;
> + }
> + }
> +}
> +
> static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
> {
> int divisor, q, r;
ok
> @@ -71,7 +114,7 @@
>
> static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
> {
> - put_bits(&s->pbctx, 3, s->channels-1); // No. of channels -1
> + put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
> put_bits(&s->pbctx, 16, 0); // Seems to be zero
> put_bits(&s->pbctx, 1, 1); // Sample count is in the header
> put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
ok
> @@ -79,6 +122,205 @@
> put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
> }
>
> +static void calc_predictor_params(AlacEncodeContext *s, int ch)
> +{
> + int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
> + int shift[MAX_LPC_ORDER];
> + int opt_order;
> +
> + opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, DEFAULT_MIN_PRED_ORDER, DEFAULT_MAX_PRED_ORDER,
> + ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
> +
> + s->lpc[ch].lpc_order = opt_order;
> + s->lpc[ch].lpc_quant = shift[opt_order-1];
> + memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
> +}
> +
I think this should be using AVCodecContext.min/max_prediction_order
> +static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
> +{
> + int i, best;
> + int32_t lt, rt;
> + uint64_t sum[4];
> + uint64_t score[4];
> +
> + /* calculate sum of 2nd order residual for each channel */
> + sum[0] = sum[1] = sum[2] = sum[3] = 0;
> + for(i=2; i<n; i++) {
> + lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
> + rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
> + sum[2] += FFABS((lt + rt) >> 1);
> + sum[3] += FFABS(lt - rt);
> + sum[0] += FFABS(lt);
> + sum[1] += FFABS(rt);
> + }
> +
> + /* calculate score for each mode */
> + score[0] = sum[0] + sum[1];
> + score[1] = sum[0] + sum[3];
> + score[2] = sum[1] + sum[3];
> + score[3] = sum[2] + sum[3];
> +
> + /* return mode with lowest score */
> + best = 0;
> + for(i=1; i<4; i++) {
> + if(score[i] < score[best]) {
> + best = i;
> + }
> + }
ok
> + if(best == 0) {
> + return ALAC_CHMODE_LEFT_RIGHT;
> + } else if(best == 1) {
> + return ALAC_CHMODE_LEFT_SIDE;
> + } else if(best == 2) {
> + return ALAC_CHMODE_RIGHT_SIDE;
> + } else {
> + return ALAC_CHMODE_MID_SIDE;
> + }
> +}
i think best could simply be returned
> +
> +static void alac_stereo_decorrelation(AlacEncodeContext *s)
> +{
> + int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
> + int i, mode, n = s->avctx->frame_size;
> +
> + mode = estimate_stereo_mode(left, right, n);
> +
> + if(mode == ALAC_CHMODE_LEFT_RIGHT) {
> + s->interlacing_leftweight = 0;
> + s->interlacing_shift = 0;
> + return;
> + }
> +
> + if(mode == ALAC_CHMODE_LEFT_SIDE) {
> + for(i=0; i<n; i++) {
> + right[i] = left[i] - right[i];
> + }
> + s->interlacing_leftweight = 1;
> + s->interlacing_shift = 0;
> +
> + } else {
> + int32_t tmp;
> + for(i=0; i<n; i++) {
> + tmp = left[i];
> + left[i] = (tmp + right[i]) >> 1;
> + right[i] = tmp - right[i];
> + }
> + s->interlacing_leftweight = 1;
> + s->interlacing_shift = 1;
> + }
i think 1 mode is missing
also it could be
if
else if
else if
else
instead of
if
return
if
else
which would be cleaner IMHO
> +}
> +
> +static void alac_linear_predictor(AlacEncodeContext *s, int ch)
> +{
> + int i;
> + LPCContext lpc = s->lpc[ch];
> +
> + if(lpc.lpc_order == 31) {
> + s->predictor_buf[0] = s->sample_buf[ch][0];
> +
> + for(i=1; i<s->avctx->frame_size; i++)
> + s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
> +
> + return;
> + }
> +
> + // generalised linear predictor
> +
> + if(lpc.lpc_order > 0) {
> + int32_t *samples = s->sample_buf[ch];
> + int32_t *residual = s->predictor_buf;
> +
> + // generate warm-up samples
> + i = lpc.lpc_order;
> + residual[0] = samples[0];
> + while(i > 0) {
> + residual[i] = samples[i] - samples[i-1];
> + i--;
> + }
this also can be changed to a for() loop
alternatively residual could be droped and the stuff could all be done
in place in samples but this may be tricky
> + // perform lpc on remaining samples
> + for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
> + int sum = 0, res_val, j;
> +
> + for (j = 0; j < lpc.lpc_order; j++) {
> + sum += (samples[lpc.lpc_order-j] - samples[0]) *
> + lpc.lpc_coeff[j];
> + }
> + sum += (1 << (lpc.lpc_quant - 1));
> + sum >>= lpc.lpc_quant;
> + sum += samples[0];
> + residual[i] = samples[lpc.lpc_order+1] - sum;
> + res_val = residual[i];
> +
> + if(res_val) {
> + int index = lpc.lpc_order - 1;
> + int neg = (res_val < 0);
> +
> + while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
> + int val = samples[0] - samples[lpc.lpc_order - index];
> + int sign = (val ? FFSIGN(val) : 0);
> +
> + if(neg)
> + sign*=-1;
> +
> + lpc.lpc_coeff[index] -= sign;
> + val *= sign;
> + res_val -= ((val >> lpc.lpc_quant) *
> + (lpc.lpc_order - index));
> + index--;
> + }
> + }
> + samples++;
> + }
> + }
> +}
> +
> +static void alac_entropy_coder(AlacEncodeContext *s)
> +{
> + unsigned int history = s->rc.initial_history;
> + int sign_modifier = 0, i, k;
> + int32_t *samples = s->predictor_buf;
> +
> + for(i=0;i < s->avctx->frame_size;) {
> + int x;
> +
> + k = av_log2((history >> 9) + 3);
> +
> + x = -2*(*samples)-1;
> + x ^= (x>>31);
> +
> + samples++;
> + i++;
> +
> + encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
> +
> + history += x * s->rc.history_mult
> + - ((history * s->rc.history_mult) >> 9);
not sure if its worth but this could be simplified to:
history -= (((history - (x<<9)) * s->rc.history_mult) >> 9);
(assuming things dont overflow)
> +
> + sign_modifier = 0;
> + if(x > 0xFFFF)
> + history = 0xFFFF;
> +
> + if((history < 128) && (i < s->avctx->frame_size)) {
> + unsigned int block_size = 0;
> +
> + sign_modifier = 1;
unused
> + k = 7 - av_log2(history) + ((history + 16) >> 6);
> +
> + while((*samples == 0) && (i < s->avctx->frame_size)) {
> + samples++;
> + i++;
> + block_size++;
> + }
> + encode_scalar(s, block_size, k, 16);
> +
> + sign_modifier = (block_size <= 0xFFFF);
> +
> + history = 0;
> + }
> +
> + }
> +}
> +
> static void write_compressed_frame(AlacEncodeContext *s)
> {
> int i, j;
> @@ -88,7 +330,7 @@
> put_bits(&s->pbctx, 8, s->interlacing_shift);
> put_bits(&s->pbctx, 8, s->interlacing_leftweight);
>
> - for(i=0;i<s->channels;i++) {
> + for(i=0;i<s->avctx->channels;i++) {
>
> calc_predictor_params(s, i);
>
> @@ -105,7 +347,7 @@
>
> // apply lpc and entropy coding to audio samples
>
> - for(i=0;i<s->channels;i++) {
> + for(i=0;i<s->avctx->channels;i++) {
> alac_linear_predictor(s, i);
> alac_entropy_coder(s);
> }
> @@ -118,8 +360,6 @@
>
> avctx->frame_size = DEFAULT_FRAME_SIZE;
> avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
> - s->channels = avctx->channels;
> - s->samplerate = avctx->sample_rate;
>
> if(avctx->sample_fmt != SAMPLE_FMT_S16) {
> av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
> @@ -139,18 +379,18 @@
> s->rc.rice_modifier = 4;
>
> s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
> - avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
> + avctx->frame_size*avctx->channels*avctx->bits_per_sample)>>3;
>
> - s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
> + s->write_sample_size = avctx->bits_per_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
>
> AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
> AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
> AV_WB32(alac_extradata+12, avctx->frame_size);
> AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
> - AV_WB8 (alac_extradata+21, s->channels);
> + AV_WB8 (alac_extradata+21, avctx->channels);
> AV_WB32(alac_extradata+24, s->max_coded_frame_size);
> - AV_WB32(alac_extradata+28, s->samplerate*s->channels*avctx->bits_per_sample); // average bitrate
> - AV_WB32(alac_extradata+32, s->samplerate);
> + AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_sample); // average bitrate
> + AV_WB32(alac_extradata+32, avctx->sample_rate);
>
> // Set relevant extradata fields
> if(s->compression_level > 0) {
> @@ -168,19 +408,66 @@
> s->avctx = avctx;
> dsputil_init(&s->dspctx, avctx);
>
> - allocate_sample_buffers(s);
> -
> return 0;
> }
>
> -static av_cold int alac_encode_close(AVCodecContext *avctx)
> +static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
> + int buf_size, void *data)
> {
> AlacEncodeContext *s = avctx->priv_data;
> + PutBitContext *pb = &s->pbctx;
> + int i, out_bytes, verbatim_flag = 0;
>
> + if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
> + av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
> + return -1;
> + }
> +
> + if(buf_size < 2*s->max_coded_frame_size) {
> + av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
> + return -1;
> + }
ok
> +
> + init_put_bits(pb, frame, buf_size);
> +
> +verbatim:
the label can be moved before the init_put_bits() which makes the second
call to it before the goto uneeded
> + if((s->compression_level == 0) || verbatim_flag) {
> + // Verbatim mode
> + int16_t *samples = data;
> + write_frame_header(s, 1);
> + for(i=0; i<avctx->frame_size*avctx->channels; i++) {
> + put_sbits(pb, 16, *samples++);
> + }
> + } else {
> + init_sample_buffers(s, data);
> + write_frame_header(s, 0);
> + write_compressed_frame(s);
> + }
> +
> + put_bits(pb, 3, 7);
> + flush_put_bits(pb);
> + out_bytes = put_bits_count(pb) >> 3;
> +
> + if(out_bytes > s->max_coded_frame_size) {
> + /* frame too large. use verbatim mode */
> + if(verbatim_flag || (s->compression_level == 0)) {
> + /* still too large. must be an error. */
> + av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
> + return -1;
> + }
ok
> + init_put_bits(pb, frame, buf_size);
> + verbatim_flag = 1;
> + goto verbatim;
> + }
> +
> + return out_bytes;
> +}
> +
> +static av_cold int alac_encode_close(AVCodecContext *avctx)
> +{
> av_freep(&avctx->extradata);
> avctx->extradata_size = 0;
> av_freep(&avctx->coded_frame);
> - free_sample_buffers(s);
> return 0;
> }
>
ok
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
The greatest way to live with honor in this world is to be what we pretend
to be. -- Socrates
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