[FFmpeg-cvslog] alacenc: store current frame size in AlacEncodeContext.
Justin Ruggles
git at videolan.org
Sun Feb 12 01:34:08 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Thu Feb 2 18:06:28 2012 -0500| [ba821b098b5748e46db0fea875679365b33110e3] | committer: Justin Ruggles
alacenc: store current frame size in AlacEncodeContext.
This avoids an indirection and will simplify implementation of encode2()
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ba821b098b5748e46db0fea875679365b33110e3
---
libavcodec/alacenc.c | 29 ++++++++++++++++-------------
1 files changed, 16 insertions(+), 13 deletions(-)
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c
index 7bc5a19..88b2f82 100644
--- a/libavcodec/alacenc.c
+++ b/libavcodec/alacenc.c
@@ -58,6 +58,7 @@ typedef struct AlacLPCContext {
} AlacLPCContext;
typedef struct AlacEncodeContext {
+ int frame_size; /**< current frame size */
int compression_level;
int min_prediction_order;
int max_prediction_order;
@@ -82,7 +83,7 @@ static void init_sample_buffers(AlacEncodeContext *s,
for (ch = 0; ch < s->avctx->channels; ch++) {
const int16_t *sptr = input_samples + ch;
- for (i = 0; i < s->avctx->frame_size; i++) {
+ for (i = 0; i < s->frame_size; i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
}
@@ -124,7 +125,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
+ put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame
}
static void calc_predictor_params(AlacEncodeContext *s, int ch)
@@ -144,7 +145,7 @@ static void calc_predictor_params(AlacEncodeContext *s, int ch)
s->lpc[ch].lpc_coeff[5] = -25;
} else {
opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
- s->avctx->frame_size,
+ s->frame_size,
s->min_prediction_order,
s->max_prediction_order,
ALAC_MAX_LPC_PRECISION, coefs, shift,
@@ -193,7 +194,7 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
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;
+ int i, mode, n = s->frame_size;
int32_t tmp;
mode = estimate_stereo_mode(left, right, n);
@@ -238,7 +239,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
if (lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
- for (i = 1; i < s->avctx->frame_size; i++) {
+ for (i = 1; i < s->frame_size; i++) {
s->predictor_buf[i] = s->sample_buf[ch][i ] -
s->sample_buf[ch][i - 1];
}
@@ -258,7 +259,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
residual[i] = samples[i] - samples[i-1];
// perform lpc on remaining samples
- for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
+ 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++) {
@@ -300,7 +301,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
int sign_modifier = 0, i, k;
int32_t *samples = s->predictor_buf;
- for (i = 0; i < s->avctx->frame_size;) {
+ for (i = 0; i < s->frame_size;) {
int x;
k = av_log2((history >> 9) + 3);
@@ -320,12 +321,12 @@ static void alac_entropy_coder(AlacEncodeContext *s)
if (x > 0xFFFF)
history = 0xFFFF;
- if (history < 128 && i < s->avctx->frame_size) {
+ 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->avctx->frame_size) {
+ while (*samples == 0 && i < s->frame_size) {
samples++;
i++;
block_size++;
@@ -369,7 +370,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
// 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->avctx->frame_size - 1; j > 0; j--)
+ for (j = s->frame_size - 1; j > 0; j--)
s->predictor_buf[j] -= s->predictor_buf[j - 1];
}
@@ -398,7 +399,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
int ret;
uint8_t *alac_extradata;
- avctx->frame_size = DEFAULT_FRAME_SIZE;
+ avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
@@ -519,8 +520,10 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int i, out_bytes, verbatim_flag = 0;
int max_frame_size;
+ s->frame_size = avctx->frame_size;
+
if (avctx->frame_size < DEFAULT_FRAME_SIZE)
- max_frame_size = get_max_frame_size(avctx->frame_size, avctx->channels,
+ max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
DEFAULT_SAMPLE_SIZE);
else
max_frame_size = s->max_coded_frame_size;
@@ -537,7 +540,7 @@ verbatim:
// Verbatim mode
const int16_t *samples = data;
write_frame_header(s, 1);
- for (i = 0; i < avctx->frame_size * avctx->channels; i++) {
+ for (i = 0; i < s->frame_size * avctx->channels; i++) {
put_sbits(pb, 16, *samples++);
}
} else {
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