*/
#include "avcodec.h"
+#include "internal.h"
#include "wma.h"
-
-#undef NDEBUG
-#include <assert.h>
+#include "libavutil/avassert.h"
static int encode_init(AVCodecContext * avctx){
return AVERROR(EINVAL);
}
+ if (avctx->sample_rate > 48000) {
+ av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz",
+ avctx->sample_rate);
+ return AVERROR(EINVAL);
+ }
+
if(avctx->bit_rate < 24*1000) {
av_log(avctx, AV_LOG_ERROR, "bitrate too low: got %i, need 24000 or higher\n",
avctx->bit_rate);
AV_WL32(extradata, flags1);
AV_WL16(extradata+4, flags2);
}else
- assert(0);
+ av_assert0(0);
avctx->extradata= extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
+ if (avctx->channels == 2)
+ s->ms_stereo = 1;
ff_wma_init(avctx, flags2);
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
- avctx->block_align=
- s->block_align= avctx->bit_rate*(int64_t)s->frame_len / (avctx->sample_rate*8);
+ s->block_align = avctx->bit_rate * (int64_t)s->frame_len /
+ (avctx->sample_rate * 8);
+ s->block_align = FFMIN(s->block_align, MAX_CODED_SUPERFRAME_SIZE);
+ avctx->block_align = s->block_align;
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
- avctx->frame_size= s->frame_len;
+ avctx->frame_size = avctx->delay = s->frame_len;
+
+#if FF_API_OLD_ENCODE_AUDIO
+ avctx->coded_frame = &s->frame;
+ avcodec_get_frame_defaults(avctx->coded_frame);
+#endif
return 0;
}
q_end = q + s->block_len;
if (s->version == 1) {
last_exp= *exp_param++;
- assert(last_exp-10 >= 0 && last_exp-10 < 32);
+ av_assert0(last_exp-10 >= 0 && last_exp-10 < 32);
put_bits(&s->pb, 5, last_exp - 10);
q+= *ptr++;
}else
while (q < q_end) {
int exp = *exp_param++;
int code = exp - last_exp + 60;
- assert(code >= 0 && code < 120);
+ av_assert1(code >= 0 && code < 120);
put_bits(&s->pb, ff_aac_scalefactor_bits[code], ff_aac_scalefactor_code[code]);
/* XXX: use a table */
q+= *ptr++;
//FIXME remove duplication relative to decoder
if (s->use_variable_block_len) {
- assert(0); //FIXME not implemented
+ av_assert0(0); //FIXME not implemented
}else{
/* fixed block len */
s->next_block_len_bits = s->frame_len_bits;
}
if (s->nb_channels == 2) {
- put_bits(&s->pb, 1, s->ms_stereo= 1);
+ put_bits(&s->pb, 1, !!s->ms_stereo);
}
for(ch = 0; ch < s->nb_channels; ch++) {
mult *= mdct_norm;
coefs = src_coefs[ch];
if (s->use_noise_coding && 0) {
- assert(0); //FIXME not implemented
+ av_assert0(0); //FIXME not implemented
} else {
coefs += s->coefs_start;
n = nb_coefs[ch];
if (s->use_exp_vlc) {
encode_exp_vlc(s, ch, fixed_exp);
} else {
- assert(0); //FIXME not implemented
+ av_assert0(0); //FIXME not implemented
// encode_exp_lsp(s, ch);
}
}
}
} else {
- assert(0); //FIXME not implemented
+ av_assert0(0); //FIXME not implemented
}
for(ch = 0; ch < s->nb_channels; ch++) {
code= run + s->int_table[tindex][abs_level-1];
}
- assert(code < s->coef_vlcs[tindex]->n);
+ av_assert2(code < s->coef_vlcs[tindex]->n);
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code], s->coef_vlcs[tindex]->huffcodes[code]);
if(code == 0){
if(1<<coef_nb_bits <= abs_level)
return -1;
-
- //Workaround minor rounding differences for the regression tests, FIXME we should find and replace the problematic float by fixpoint for reg tests
- if(abs_level == 0x71B && (s->avctx->flags & CODEC_FLAG_BITEXACT)) abs_level=0x71A;
-
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, run);
}
init_put_bits(&s->pb, buf, buf_size);
if (s->use_bit_reservoir) {
- assert(0);//FIXME not implemented
+ av_assert0(0);//FIXME not implemented
}else{
if(encode_block(s, src_coefs, total_gain) < 0)
return INT_MAX;
return put_bits_count(&s->pb)/8 - s->block_align;
}
-static int encode_superframe(AVCodecContext *avctx,
- unsigned char *buf, int buf_size, void *data){
+static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
WMACodecContext *s = avctx->priv_data;
- const short *samples = data;
- int i, total_gain;
+ const int16_t *samples = (const int16_t *)frame->data[0];
+ int i, total_gain, ret, error;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits;
- apply_window_and_mdct(avctx, samples, avctx->frame_size);
+ apply_window_and_mdct(avctx, samples, frame->nb_samples);
if (s->ms_stereo) {
float a, b;
}
}
-#if 1
+ if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE)))
+ return ret;
+
total_gain= 128;
for(i=64; i; i>>=1){
- int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
- if(error<0)
+ error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
+ total_gain - i);
+ if(error<=0)
total_gain-= i;
}
-#else
- total_gain= 90;
- best= encode_frame(s, s->coefs, buf, buf_size, total_gain);
- for(i=32; i; i>>=1){
- int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
- int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
- av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
- if(scoreL < FFMIN(best, scoreR)){
- best = scoreL;
- total_gain -= i;
- }else if(scoreR < best){
- best = scoreR;
- total_gain += i;
- }
- }
-#endif
- encode_frame(s, s->coefs, buf, buf_size, total_gain);
- assert((put_bits_count(&s->pb) & 7) == 0);
+ while(total_gain <= 128 && error > 0)
+ error = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain++);
+ av_assert0((put_bits_count(&s->pb) & 7) == 0);
i= s->block_align - (put_bits_count(&s->pb)+7)/8;
- assert(i>=0);
+ av_assert0(i>=0);
while(i--)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
- return put_bits_ptr(&s->pb) - s->pb.buf;
+ av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == s->block_align);
+
+ if (frame->pts != AV_NOPTS_VALUE)
+ avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+ avpkt->size = s->block_align;
+ *got_packet_ptr = 1;
+ return 0;
}
+#if CONFIG_WMAV1_ENCODER
AVCodec ff_wmav1_encoder = {
.name = "wmav1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
- .encode = encode_superframe,
+ .encode2 = encode_superframe,
.close = ff_wma_end,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
};
-
+#endif
+#if CONFIG_WMAV2_ENCODER
AVCodec ff_wmav2_encoder = {
.name = "wmav2",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
- .encode = encode_superframe,
+ .encode2 = encode_superframe,
.close = ff_wma_end,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};
+#endif
projects / ffmpeg / blobdiff