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Import more ok'd parts of ALAC encoder from GSoC repo.

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Originally committed as revision 14820 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Ramiro Polla 2008-08-17 22:47:40 +00:00
parent 46dd2738ae
commit ca04826627
1 changed files with 125 additions and 15 deletions
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140
libavcodec/alacenc.c
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@ -33,15 +33,52 @@
#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
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];
int interlacing_shift;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
LPCContext lpc[MAX_CHANNELS];
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;
@ -71,7 +108,7 @@ static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_s
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
@ -79,6 +116,38 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
}
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;
}
}
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
@ -88,7 +157,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
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 +174,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
// 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 +187,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
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 +206,18 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
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 +235,62 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->avctx = avctx;
dsputil_init(&s->dspctx, avctx);
allocate_sample_buffers(s);
return 0;
}
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;
}
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;
}
verbatim_flag = 1;
goto verbatim;
}
return out_bytes;
}
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
free_sample_buffers(s);
return 0;
}