[FFmpeg-cvslog] alacenc: pretty-printing and other cosmetics
Justin Ruggles
git at videolan.org
Sun Feb 12 01:34:07 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Wed Feb 1 21:21:24 2012 -0500| [fc9cf0b2a6a0bd3933fcef216860c594b767834e] | committer: Justin Ruggles
alacenc: pretty-printing and other cosmetics
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=fc9cf0b2a6a0bd3933fcef216860c594b767834e
---
libavcodec/alacenc.c | 135 ++++++++++++++++++++++++--------------------------
1 files changed, 64 insertions(+), 71 deletions(-)
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c
index 934541e..9e632ae 100644
--- a/libavcodec/alacenc.c
+++ b/libavcodec/alacenc.c
@@ -119,12 +119,12 @@ static void encode_scalar(AlacEncodeContext *s, int x,
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
{
- 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
- 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_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
+ 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
}
static void calc_predictor_params(AlacEncodeContext *s, int ch)
@@ -167,8 +167,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
/* 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];
+ 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);
@@ -184,9 +184,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
/* return mode with lowest score */
best = 0;
for (i = 1; i < 4; i++) {
- if (score[i] < score[best]) {
+ if (score[i] < score[best])
best = i;
- }
}
return best;
}
@@ -199,40 +198,35 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s)
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;
+ 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;
}
}
@@ -244,8 +238,10 @@ 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++)
- s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
+ 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;
}
@@ -267,7 +263,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
for (j = 0; j < lpc.lpc_order; j++) {
sum += (samples[lpc.lpc_order-j] - samples[0]) *
- lpc.lpc_coeff[j];
+ lpc.lpc_coeff[j];
}
sum >>= lpc.lpc_quant;
@@ -276,21 +272,20 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
s->write_sample_size);
res_val = residual[i];
- if(res_val) {
+ 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];
+ 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;
+ if (neg)
+ sign *= -1;
lpc.lpc_coeff[index] -= sign;
val *= sign;
- res_val -= ((val >> lpc.lpc_quant) *
- (lpc.lpc_order - index));
+ res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);
index--;
}
}
@@ -310,16 +305,16 @@ static void alac_entropy_coder(AlacEncodeContext *s)
k = av_log2((history >> 9) + 3);
- x = -2*(*samples)-1;
- x ^= (x>>31);
+ 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);
+ history += x * s->rc.history_mult -
+ ((history * s->rc.history_mult) >> 9);
sign_modifier = 0;
if (x > 0xFFFF)
@@ -336,9 +331,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
block_size++;
}
encode_scalar(s, block_size, k, 16);
-
sign_modifier = (block_size <= 0xFFFF);
-
history = 0;
}
@@ -356,7 +349,6 @@ static void write_compressed_frame(AlacEncodeContext *s)
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
for (i = 0; i < s->avctx->channels; i++) {
-
calc_predictor_params(s, i);
put_bits(&s->pbctx, 4, prediction_type);
@@ -365,9 +357,8 @@ static void write_compressed_frame(AlacEncodeContext *s)
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++) {
+ for (j = 0; j < s->lpc[i].lpc_order; j++)
put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
- }
}
// apply lpc and entropy coding to audio samples
@@ -398,11 +389,11 @@ static av_cold int alac_encode_close(AVCodecContext *avctx)
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
- AlacEncodeContext *s = avctx->priv_data;
+ AlacEncodeContext *s = avctx->priv_data;
int ret;
uint8_t *alac_extradata;
- avctx->frame_size = DEFAULT_FRAME_SIZE;
+ avctx->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");
@@ -429,9 +420,11 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->rc.k_modifier = 14;
s->rc.rice_modifier = 4;
- s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels * DEFAULT_SAMPLE_SIZE >> 3);
+ s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels *
+ DEFAULT_SAMPLE_SIZE >> 3);
- s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes
+ // FIXME: consider wasted_bytes
+ s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1;
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) {
@@ -566,8 +559,8 @@ AVCodec ff_alac_encoder = {
.init = alac_encode_init,
.encode = alac_encode_frame,
.close = alac_encode_close,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_NONE },
- .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
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