-/**
+/*
* ALAC audio encoder
* Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
-#include "bitstream.h"
-#include "dsputil.h"
+#include "put_bits.h"
+#include "internal.h"
#include "lpc.h"
+#include "mathops.h"
+#include "alac_data.h"
#define DEFAULT_FRAME_SIZE 4096
-#define DEFAULT_SAMPLE_SIZE 16
-#define MAX_CHANNELS 8
#define ALAC_EXTRADATA_SIZE 36
#define ALAC_FRAME_HEADER_SIZE 55
#define ALAC_FRAME_FOOTER_SIZE 3
#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
+
+#define ALAC_CHMODE_LEFT_RIGHT 0
+#define ALAC_CHMODE_LEFT_SIDE 1
+#define ALAC_CHMODE_RIGHT_SIDE 2
+#define ALAC_CHMODE_MID_SIDE 3
+
+typedef struct RiceContext {
+ int history_mult;
+ int initial_history;
+ int k_modifier;
+ int rice_modifier;
+} RiceContext;
+
+typedef struct AlacLPCContext {
+ int lpc_order;
+ int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
+ int lpc_quant;
+} AlacLPCContext;
+
+typedef struct AlacEncodeContext {
+ int frame_size; /**< current frame size */
+ int verbatim; /**< current frame verbatim mode flag */
+ int compression_level;
+ int min_prediction_order;
+ int max_prediction_order;
+ int max_coded_frame_size;
+ int write_sample_size;
+ int extra_bits;
+ int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
+ int32_t predictor_buf[DEFAULT_FRAME_SIZE];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
- DSPContext dspctx;
+ RiceContext rc;
+ AlacLPCContext lpc[2];
+ LPCContext lpc_ctx;
AVCodecContext *avctx;
} AlacEncodeContext;
-static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
+static void init_sample_buffers(AlacEncodeContext *s, int channels,
+ uint8_t const *samples[2])
+{
+ int ch, i;
+ int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
+ s->avctx->bits_per_raw_sample;
+
+#define COPY_SAMPLES(type) do { \
+ for (ch = 0; ch < channels; ch++) { \
+ int32_t *bptr = s->sample_buf[ch]; \
+ const type *sptr = (const type *)samples[ch]; \
+ for (i = 0; i < s->frame_size; i++) \
+ bptr[i] = sptr[i] >> shift; \
+ } \
+ } while (0)
+
+ if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
+ COPY_SAMPLES(int32_t);
+ else
+ COPY_SAMPLES(int16_t);
+}
+
+static void encode_scalar(AlacEncodeContext *s, int x,
+ int k, int write_sample_size)
{
int divisor, q, r;
q = x / divisor;
r = x % divisor;
- if(q > 8) {
+ if (q > 8) {
// write escape code and sample value directly
put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
put_bits(&s->pbctx, write_sample_size, x);
} else {
- if(q)
+ if (q)
put_bits(&s->pbctx, q, (1<<q) - 1);
put_bits(&s->pbctx, 1, 0);
- if(k != 1) {
- if(r > 0)
+ if (k != 1) {
+ if (r > 0)
put_bits(&s->pbctx, k, r+1);
else
put_bits(&s->pbctx, k-1, 0);
}
}
-static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
+static void write_element_header(AlacEncodeContext *s,
+ enum AlacRawDataBlockType element,
+ int instance)
{
- put_bits(&s->pbctx, 3, s->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
- put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
- put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
+ int encode_fs = 0;
+
+ if (s->frame_size < DEFAULT_FRAME_SIZE)
+ encode_fs = 1;
+
+ put_bits(&s->pbctx, 3, element); // element type
+ put_bits(&s->pbctx, 4, instance); // element instance
+ put_bits(&s->pbctx, 12, 0); // unused header bits
+ put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
+ put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
+ put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
+ if (encode_fs)
+ put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
}
-static void write_compressed_frame(AlacEncodeContext *s)
+static void calc_predictor_params(AlacEncodeContext *s, int ch)
{
- int i, j;
+ int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
+ int shift[MAX_LPC_ORDER];
+ int opt_order;
+
+ if (s->compression_level == 1) {
+ s->lpc[ch].lpc_order = 6;
+ s->lpc[ch].lpc_quant = 6;
+ s->lpc[ch].lpc_coeff[0] = 160;
+ s->lpc[ch].lpc_coeff[1] = -190;
+ s->lpc[ch].lpc_coeff[2] = 170;
+ s->lpc[ch].lpc_coeff[3] = -130;
+ s->lpc[ch].lpc_coeff[4] = 80;
+ s->lpc[ch].lpc_coeff[5] = -25;
+ } else {
+ opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
+ s->frame_size,
+ s->min_prediction_order,
+ s->max_prediction_order,
+ ALAC_MAX_LPC_PRECISION, coefs, shift,
+ FF_LPC_TYPE_LEVINSON, 0,
+ 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));
+ }
+}
- /* only simple mid/side decorrelation supported as of now */
- alac_stereo_decorrelation(s);
- put_bits(&s->pbctx, 8, s->interlacing_shift);
- put_bits(&s->pbctx, 8, s->interlacing_leftweight);
+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);
+ }
- for(i=0;i<s->channels;i++) {
+ /* 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;
+ }
+ return best;
+}
- calc_predictor_params(s, i);
+static void alac_stereo_decorrelation(AlacEncodeContext *s)
+{
+ int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
+ int i, mode, n = s->frame_size;
+ int32_t tmp;
+
+ mode = estimate_stereo_mode(left, right, n);
+
+ switch (mode) {
+ case ALAC_CHMODE_LEFT_RIGHT:
+ s->interlacing_leftweight = 0;
+ s->interlacing_shift = 0;
+ break;
+ case ALAC_CHMODE_LEFT_SIDE:
+ for (i = 0; i < n; i++)
+ right[i] = left[i] - right[i];
+ s->interlacing_leftweight = 1;
+ s->interlacing_shift = 0;
+ break;
+ case ALAC_CHMODE_RIGHT_SIDE:
+ for (i = 0; i < n; i++) {
+ tmp = right[i];
+ right[i] = left[i] - right[i];
+ left[i] = tmp + (right[i] >> 31);
+ }
+ s->interlacing_leftweight = 1;
+ s->interlacing_shift = 31;
+ break;
+ default:
+ 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;
+ break;
+ }
+}
+
+static void alac_linear_predictor(AlacEncodeContext *s, int ch)
+{
+ int i;
+ AlacLPCContext lpc = s->lpc[ch];
- put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
- put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
+ if (lpc.lpc_order == 31) {
+ s->predictor_buf[0] = s->sample_buf[ch][0];
- put_bits(&s->pbctx, 3, s->rc.rice_modifier);
- put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
- // predictor coeff. table
- for(j=0;j<s->lpc[i].lpc_order;j++) {
- put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
+ for (i = 1; i < s->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
+ residual[0] = samples[0];
+ for (i = 1; i <= lpc.lpc_order; i++)
+ residual[i] = samples[i] - samples[i-1];
+
+ // perform lpc on remaining samples
+ for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {
+ int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
+
+ for (j = 0; j < lpc.lpc_order; j++) {
+ sum += (samples[lpc.lpc_order-j] - samples[0]) *
+ lpc.lpc_coeff[j];
+ }
+
+ sum >>= lpc.lpc_quant;
+ sum += samples[0];
+ residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
+ s->write_sample_size);
+ 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++;
}
}
+}
- // apply lpc and entropy coding to audio 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->channels;i++) {
- alac_linear_predictor(s, i);
- alac_entropy_coder(s);
+ for (i = 0; i < s->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);
+
+ sign_modifier = 0;
+ if (x > 0xFFFF)
+ history = 0xFFFF;
+
+ if (history < 128 && i < s->frame_size) {
+ unsigned int block_size = 0;
+
+ k = 7 - av_log2(history) + ((history + 16) >> 6);
+
+ while (*samples == 0 && i < s->frame_size) {
+ samples++;
+ i++;
+ block_size++;
+ }
+ encode_scalar(s, block_size, k, 16);
+ sign_modifier = (block_size <= 0xFFFF);
+ history = 0;
+ }
+
+ }
+}
+
+static void write_element(AlacEncodeContext *s,
+ enum AlacRawDataBlockType element, int instance,
+ const uint8_t *samples0, const uint8_t *samples1)
+{
+ uint8_t const *samples[2] = { samples0, samples1 };
+ int i, j, channels;
+ int prediction_type = 0;
+ PutBitContext *pb = &s->pbctx;
+
+ channels = element == TYPE_CPE ? 2 : 1;
+
+ if (s->verbatim) {
+ write_element_header(s, element, instance);
+ /* samples are channel-interleaved in verbatim mode */
+ if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
+ int shift = 32 - s->avctx->bits_per_raw_sample;
+ int32_t const *samples_s32[2] = { (const int32_t *)samples0,
+ (const int32_t *)samples1 };
+ for (i = 0; i < s->frame_size; i++)
+ for (j = 0; j < channels; j++)
+ put_sbits(pb, s->avctx->bits_per_raw_sample,
+ samples_s32[j][i] >> shift);
+ } else {
+ int16_t const *samples_s16[2] = { (const int16_t *)samples0,
+ (const int16_t *)samples1 };
+ for (i = 0; i < s->frame_size; i++)
+ for (j = 0; j < channels; j++)
+ put_sbits(pb, s->avctx->bits_per_raw_sample,
+ samples_s16[j][i]);
+ }
+ } else {
+ s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
+ channels - 1;
+
+ init_sample_buffers(s, channels, samples);
+ write_element_header(s, element, instance);
+
+ if (channels == 2)
+ alac_stereo_decorrelation(s);
+ else
+ s->interlacing_shift = s->interlacing_leftweight = 0;
+ put_bits(pb, 8, s->interlacing_shift);
+ put_bits(pb, 8, s->interlacing_leftweight);
+
+ for (i = 0; i < channels; i++) {
+ calc_predictor_params(s, i);
+
+ put_bits(pb, 4, prediction_type);
+ put_bits(pb, 4, s->lpc[i].lpc_quant);
+
+ put_bits(pb, 3, s->rc.rice_modifier);
+ put_bits(pb, 5, s->lpc[i].lpc_order);
+ // predictor coeff. table
+ for (j = 0; j < s->lpc[i].lpc_order; j++)
+ put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
+ }
+
+ // write extra bits if needed
+ if (s->extra_bits) {
+ uint32_t mask = (1 << s->extra_bits) - 1;
+ for (i = 0; i < s->frame_size; i++) {
+ for (j = 0; j < channels; j++) {
+ put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask);
+ s->sample_buf[j][i] >>= s->extra_bits;
+ }
+ }
+ }
+
+ // apply lpc and entropy coding to audio samples
+ for (i = 0; i < channels; i++) {
+ alac_linear_predictor(s, i);
+
+ // TODO: determine when this will actually help. for now it's not used.
+ if (prediction_type == 15) {
+ // 2nd pass 1st order filter
+ for (j = s->frame_size - 1; j > 0; j--)
+ s->predictor_buf[j] -= s->predictor_buf[j - 1];
+ }
+ alac_entropy_coder(s);
+ }
}
}
+static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
+ uint8_t * const *samples)
+{
+ PutBitContext *pb = &s->pbctx;
+ const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
+ const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
+ int ch, element, sce, cpe;
+
+ init_put_bits(pb, avpkt->data, avpkt->size);
+
+ ch = element = sce = cpe = 0;
+ while (ch < s->avctx->channels) {
+ if (ch_elements[element] == TYPE_CPE) {
+ write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
+ samples[ch_map[ch + 1]]);
+ cpe++;
+ ch += 2;
+ } else {
+ write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
+ sce++;
+ ch++;
+ }
+ element++;
+ }
+
+ put_bits(pb, 3, TYPE_END);
+ flush_put_bits(pb);
+
+ return put_bits_count(pb) >> 3;
+}
+
+static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
+{
+ int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
+ return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;
+}
+
+static av_cold int alac_encode_close(AVCodecContext *avctx)
+{
+ AlacEncodeContext *s = avctx->priv_data;
+ ff_lpc_end(&s->lpc_ctx);
+ av_freep(&avctx->extradata);
+ avctx->extradata_size = 0;
+ av_freep(&avctx->coded_frame);
+ return 0;
+}
+
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
- AlacEncodeContext *s = avctx->priv_data;
- uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
+ AlacEncodeContext *s = avctx->priv_data;
+ int ret;
+ uint8_t *alac_extradata;
- avctx->frame_size = DEFAULT_FRAME_SIZE;
- avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
- s->channels = avctx->channels;
- s->samplerate = avctx->sample_rate;
+ avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
- if(avctx->sample_fmt != SAMPLE_FMT_S16) {
- av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
- return -1;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
+ if (avctx->bits_per_raw_sample != 24)
+ av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
+ avctx->bits_per_raw_sample = 24;
+ } else {
+ avctx->bits_per_raw_sample = 16;
+ s->extra_bits = 0;
}
// Set default compression level
- if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
- s->compression_level = 1;
+ if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
+ s->compression_level = 2;
else
- s->compression_level = av_clip(avctx->compression_level, 0, 1);
+ s->compression_level = av_clip(avctx->compression_level, 0, 2);
// Initialize default Rice parameters
s->rc.history_mult = 40;
s->rc.k_modifier = 14;
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;
+ s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,
+ avctx->channels,
+ avctx->bits_per_raw_sample);
- s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
+ avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+ avctx->extradata_size = ALAC_EXTRADATA_SIZE;
+ alac_extradata = avctx->extradata;
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+17, avctx->bits_per_raw_sample);
+ 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_raw_sample); // average bitrate
+ AV_WB32(alac_extradata+32, avctx->sample_rate);
// Set relevant extradata fields
- if(s->compression_level > 0) {
+ if (s->compression_level > 0) {
AV_WB8(alac_extradata+18, s->rc.history_mult);
AV_WB8(alac_extradata+19, s->rc.initial_history);
AV_WB8(alac_extradata+20, s->rc.k_modifier);
}
- avctx->extradata = alac_extradata;
- avctx->extradata_size = ALAC_EXTRADATA_SIZE;
+ s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
+ if (avctx->min_prediction_order >= 0) {
+ if (avctx->min_prediction_order < MIN_LPC_ORDER ||
+ avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
+ av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
+ avctx->min_prediction_order);
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ s->min_prediction_order = avctx->min_prediction_order;
+ }
+
+ s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
+ if (avctx->max_prediction_order >= 0) {
+ if (avctx->max_prediction_order < MIN_LPC_ORDER ||
+ avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
+ av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
+ avctx->max_prediction_order);
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ s->max_prediction_order = avctx->max_prediction_order;
+ }
+
+ if (s->max_prediction_order < s->min_prediction_order) {
+ av_log(avctx, AV_LOG_ERROR,
+ "invalid prediction orders: min=%d max=%d\n",
+ s->min_prediction_order, s->max_prediction_order);
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
- avctx->coded_frame = avcodec_alloc_frame();
- avctx->coded_frame->key_frame = 1;
+ avctx->coded_frame = av_frame_alloc();
+ if (!avctx->coded_frame) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
s->avctx = avctx;
- dsputil_init(&s->dspctx, avctx);
- allocate_sample_buffers(s);
+ if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
+ s->max_prediction_order,
+ FF_LPC_TYPE_LEVINSON)) < 0) {
+ goto error;
+ }
return 0;
+error:
+ alac_encode_close(avctx);
+ return ret;
}
-static av_cold int alac_encode_close(AVCodecContext *avctx)
+static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
AlacEncodeContext *s = avctx->priv_data;
+ int out_bytes, max_frame_size, ret;
- av_freep(&avctx->extradata);
- avctx->extradata_size = 0;
- av_freep(&avctx->coded_frame);
- free_sample_buffers(s);
+ s->frame_size = frame->nb_samples;
+
+ if (frame->nb_samples < DEFAULT_FRAME_SIZE)
+ max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
+ avctx->bits_per_raw_sample);
+ else
+ max_frame_size = s->max_coded_frame_size;
+
+ if ((ret = ff_alloc_packet(avpkt, 2 * max_frame_size))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+
+ /* use verbatim mode for compression_level 0 */
+ if (s->compression_level) {
+ s->verbatim = 0;
+ s->extra_bits = avctx->bits_per_raw_sample - 16;
+ } else {
+ s->verbatim = 1;
+ s->extra_bits = 0;
+ }
+
+ out_bytes = write_frame(s, avpkt, frame->extended_data);
+
+ if (out_bytes > max_frame_size) {
+ /* frame too large. use verbatim mode */
+ s->verbatim = 1;
+ s->extra_bits = 0;
+ out_bytes = write_frame(s, avpkt, frame->extended_data);
+ }
+
+ avpkt->size = out_bytes;
+ *got_packet_ptr = 1;
return 0;
}
-AVCodec alac_encoder = {
- "alac",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ALAC,
- sizeof(AlacEncodeContext),
- alac_encode_init,
- alac_encode_frame,
- alac_encode_close,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
- .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+AVCodec ff_alac_encoder = {
+ .name = "alac",
+ .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_ALAC,
+ .priv_data_size = sizeof(AlacEncodeContext),
+ .init = alac_encode_init,
+ .encode2 = alac_encode_frame,
+ .close = alac_encode_close,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .channel_layouts = ff_alac_channel_layouts,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff