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alacenc: consolidate bitstream writing into a single function.

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Simplifies use of verbatim mode.
This commit is contained in:
Justin Ruggles 2012-02-03 17:53:41 -05:00
parent b590f3a7bf
commit b6e8ff72ea
1 changed files with 33 additions and 38 deletions
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71
libavcodec/alacenc.c
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@ -59,6 +59,7 @@ typedef struct AlacLPCContext {
typedef struct AlacEncodeContext { typedef struct AlacEncodeContext {
int frame_size; /**< current frame size */ int frame_size; /**< current frame size */
int verbatim; /**< current frame verbatim mode flag */
int compression_level; int compression_level;
int min_prediction_order; int min_prediction_order;
int max_prediction_order; int max_prediction_order;
@ -118,7 +119,7 @@ static void encode_scalar(AlacEncodeContext *s, int x,
} }
} }
static void write_frame_header(AlacEncodeContext *s, int is_verbatim) static void write_frame_header(AlacEncodeContext *s)
{ {
int encode_fs = 0; int encode_fs = 0;
@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
put_bits(&s->pbctx, 16, 0); // Seems to be zero put_bits(&s->pbctx, 16, 0); // Seems to be zero
put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field 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, 1, s->verbatim); // Audio block is verbatim
if (encode_fs) if (encode_fs)
put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
} }
@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s)
} }
} }
static void write_compressed_frame(AlacEncodeContext *s) static int write_frame(AlacEncodeContext *s, uint8_t *data, int size,
const int16_t *samples)
{ {
int i, j; int i, j;
int prediction_type = 0; int prediction_type = 0;
PutBitContext *pb = &s->pbctx;
init_put_bits(pb, data, size);
if (s->verbatim) {
write_frame_header(s);
for (i = 0; i < s->frame_size * s->avctx->channels; i++)
put_sbits(pb, 16, *samples++);
} else {
init_sample_buffers(s, samples);
write_frame_header(s);
if (s->avctx->channels == 2) if (s->avctx->channels == 2)
alac_stereo_decorrelation(s); alac_stereo_decorrelation(s);
put_bits(&s->pbctx, 8, s->interlacing_shift); put_bits(pb, 8, s->interlacing_shift);
put_bits(&s->pbctx, 8, s->interlacing_leftweight); put_bits(pb, 8, s->interlacing_leftweight);
for (i = 0; i < s->avctx->channels; i++) { for (i = 0; i < s->avctx->channels; i++) {
calc_predictor_params(s, i); calc_predictor_params(s, i);
put_bits(&s->pbctx, 4, prediction_type); put_bits(pb, 4, prediction_type);
put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant); put_bits(pb, 4, s->lpc[i].lpc_quant);
put_bits(&s->pbctx, 3, s->rc.rice_modifier); put_bits(pb, 3, s->rc.rice_modifier);
put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); put_bits(pb, 5, s->lpc[i].lpc_order);
// predictor coeff. table // predictor coeff. table
for (j = 0; j < s->lpc[i].lpc_order; j++) for (j = 0; j < s->lpc[i].lpc_order; j++)
put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]); put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);
} }
// apply lpc and entropy coding to audio samples // apply lpc and entropy coding to audio samples
@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *s)
alac_entropy_coder(s); alac_entropy_coder(s);
} }
}
put_bits(pb, 3, 7);
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) static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data) int buf_size, void *data)
{ {
AlacEncodeContext *s = avctx->priv_data; AlacEncodeContext *s = avctx->priv_data;
PutBitContext *pb = &s->pbctx; int out_bytes, max_frame_size;
int i, out_bytes, verbatim_flag = 0;
int max_frame_size;
s->frame_size = avctx->frame_size; s->frame_size = avctx->frame_size;
@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
return AVERROR(EINVAL); return AVERROR(EINVAL);
} }
verbatim: /* use verbatim mode for compression_level 0 */
init_put_bits(pb, frame, buf_size); s->verbatim = !s->compression_level;
if (s->compression_level == 0 || verbatim_flag) { out_bytes = write_frame(s, frame, buf_size, data);
// Verbatim mode
const int16_t *samples = data;
write_frame_header(s, 1);
for (i = 0; i < s->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 > max_frame_size) { if (out_bytes > max_frame_size) {
/* frame too large. use verbatim mode */ /* frame too large. use verbatim mode */
if (verbatim_flag || s->compression_level == 0) { s->verbatim = 1;
/* still too large. must be an error. */ out_bytes = write_frame(s, frame, buf_size, data);
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
return AVERROR_BUG;
}
verbatim_flag = 1;
goto verbatim;
} }
return out_bytes; return out_bytes;