[FFmpeg-cvslog] g732_1: reduce difference to qatar
Michael Niedermayer
git at videolan.org
Sat Aug 4 00:30:58 CEST 2012
ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Fri Aug 3 23:50:09 2012 +0200| [70bcdfb39f1b28a7762af2aa82524d16d1128798] | committer: Michael Niedermayer
g732_1: reduce difference to qatar
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=70bcdfb39f1b28a7762af2aa82524d16d1128798
---
libavcodec/g723_1.c | 42 +++++++++++++++++++++++++-----------------
1 file changed, 25 insertions(+), 17 deletions(-)
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
index 80d1a04..155dcc7 100644
--- a/libavcodec/g723_1.c
+++ b/libavcodec/g723_1.c
@@ -65,8 +65,8 @@ typedef struct g723_1_context {
int reflection_coef;
int pf_gain; ///< formant postfilter
///< gain scaling unit memory
-
int postfilter;
+ int16_t audio[FRAME_LEN + LPC_ORDER];
int16_t prev_data[HALF_FRAME_LEN];
int16_t prev_weight_sig[PITCH_MAX];
@@ -982,7 +982,11 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
int16_t *out;
int bad_frame = 0, i, j, ret;
- if (!buf_size || buf_size < frame_size[dec_mode]) {
+ if (buf_size < frame_size[dec_mode]) {
+ if (buf_size)
+ av_log(avctx, AV_LOG_WARNING,
+ "Expected %d bytes, got %d - skipping packet\n",
+ frame_size[dec_mode], buf_size);
*got_frame_ptr = 0;
return buf_size;
}
@@ -995,7 +999,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
p->cur_frame_type = UNTRANSMITTED_FRAME;
}
- p->frame.nb_samples = FRAME_LEN + LPC_ORDER;
+ p->frame.nb_samples = FRAME_LEN;
if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
@@ -1041,7 +1045,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
vector_ptr = p->excitation + PITCH_MAX;
/* Save the excitation */
- memcpy(out, vector_ptr, FRAME_LEN * sizeof(int16_t));
+ memcpy(p->audio + LPC_ORDER, vector_ptr, FRAME_LEN * sizeof(*p->audio));
p->interp_index = comp_interp_index(p, p->pitch_lag[1],
&p->sid_gain, &p->cur_gain);
@@ -1056,27 +1060,29 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
/* Restore the original excitation */
memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
- memcpy(vector_ptr, out, FRAME_LEN * sizeof(*vector_ptr));
+ memcpy(vector_ptr, p->audio + LPC_ORDER, FRAME_LEN * sizeof(*vector_ptr));
/* Peform pitch postfiltering */
if (p->postfilter)
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_acelp_weighted_vector_sum(out + LPC_ORDER + i,
+ ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
vector_ptr + i,
vector_ptr + i + ppf[j].index,
ppf[j].sc_gain,
ppf[j].opt_gain,
1 << 14, 15, SUBFRAME_LEN);
+
} else {
p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
if (p->erased_frames == 3) {
/* Mute output */
memset(p->excitation, 0,
(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
- memset(out, 0, (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
+ memset(p->frame.data[0], 0,
+ (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
} else {
/* Regenerate frame */
- residual_interp(p->excitation, out + LPC_ORDER, p->interp_index,
+ residual_interp(p->excitation, p->audio + LPC_ORDER, p->interp_index,
p->interp_gain, &p->random_seed);
}
}
@@ -1087,28 +1093,29 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
memset(out, 0, FRAME_LEN * 2);
av_log(avctx, AV_LOG_WARNING,
"G.723.1: Comfort noise generation not supported yet\n");
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = p->frame;
return frame_size[dec_mode];
}
p->past_frame_type = p->cur_frame_type;
- memcpy(out, p->synth_mem, LPC_ORDER * sizeof(int16_t));
+ memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_celp_lp_synthesis_filter(out + i, &lpc[j * LPC_ORDER],
- out + i, SUBFRAME_LEN, LPC_ORDER,
+ ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
+ p->audio + i, SUBFRAME_LEN, LPC_ORDER,
0, 1, 1 << 12);
- memcpy(p->synth_mem, out + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
+ memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
if (p->postfilter) {
- formant_postfilter(p, lpc, out);
+ formant_postfilter(p, lpc, p->audio);
+ memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2);
} else { // if output is not postfiltered it should be scaled by 2
for (i = 0; i < FRAME_LEN; i++)
- out[LPC_ORDER + i] = av_clip_int16(out[LPC_ORDER + i] << 1);
+ out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
}
- memmove(out, out + LPC_ORDER, sizeof(int16_t)*FRAME_LEN);
- p->frame.nb_samples = FRAME_LEN;
-
*got_frame_ptr = 1;
*(AVFrame *)data = p->frame;
@@ -1124,6 +1131,7 @@ static const AVOption options[] = {
{ NULL }
};
+
static const AVClass g723_1dec_class = {
.class_name = "G.723.1 decoder",
.item_name = av_default_item_name,
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