virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

wmaenc: use AVCodec.encode2()

Browse Source
This commit is contained in:
Justin Ruggles 2012-02-20 12:29:14 -05:00
parent b0f75ba272
commit 27bacfeb57
1 changed files with 31 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

48
libavcodec/wmaenc.c
View File

@ -20,6 +20,7 @@
*/ */
#include "avcodec.h" #include "avcodec.h"
#include "internal.h"
#include "wma.h" #include "wma.h"
#undef NDEBUG #undef NDEBUG
@ -86,7 +87,12 @@ static int encode_init(AVCodecContext * avctx){
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate / avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len; s->frame_len;
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate); //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; return 0;
} }
@ -340,16 +346,17 @@ static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
return put_bits_count(&s->pb)/8 - s->block_align; return put_bits_count(&s->pb)/8 - s->block_align;
} }
static int encode_superframe(AVCodecContext *avctx, static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
unsigned char *buf, int buf_size, void *data){ const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data; WMACodecContext *s = avctx->priv_data;
const short *samples = data; const int16_t *samples = (const int16_t *)frame->data[0];
int i, total_gain; int i, total_gain, ret;
s->block_len_bits= s->frame_len_bits; //required by non variable block len s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits; 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) { if (s->ms_stereo) {
float a, b; float a, b;
@ -363,24 +370,25 @@ static int encode_superframe(AVCodecContext *avctx,
} }
} }
if (buf_size < 2 * MAX_CODED_SUPERFRAME_SIZE) { if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small\n"); av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return AVERROR(EINVAL); return ret;
} }
#if 1 #if 1
total_gain= 128; total_gain= 128;
for(i=64; i; i>>=1){ for(i=64; i; i>>=1){
int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i); int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if(error<0) if(error<0)
total_gain-= i; total_gain-= i;
} }
#else #else
total_gain= 90; total_gain= 90;
best= encode_frame(s, s->coefs, buf, buf_size, total_gain); best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for(i=32; i; i>>=1){ for(i=32; i; i>>=1){
int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i); int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i); int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain); av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if(scoreL < FFMIN(best, scoreR)){ if(scoreL < FFMIN(best, scoreR)){
best = scoreL; best = scoreL;
@ -392,7 +400,7 @@ static int encode_superframe(AVCodecContext *avctx,
} }
#endif #endif
if ((i = encode_frame(s, s->coefs, buf, buf_size, total_gain)) >= 0) { if ((i = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain)) >= 0) {
av_log(avctx, AV_LOG_ERROR, "required frame size too large. please " av_log(avctx, AV_LOG_ERROR, "required frame size too large. please "
"use a higher bit rate.\n"); "use a higher bit rate.\n");
return AVERROR(EINVAL); return AVERROR(EINVAL);
@ -402,7 +410,13 @@ static int encode_superframe(AVCodecContext *avctx,
put_bits(&s->pb, 8, 'N'); put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb); flush_put_bits(&s->pb);
return 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;
} }
AVCodec ff_wmav1_encoder = { AVCodec ff_wmav1_encoder = {
@ -411,7 +425,7 @@ AVCodec ff_wmav1_encoder = {
.id = CODEC_ID_WMAV1, .id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext), .priv_data_size = sizeof(WMACodecContext),
.init = encode_init, .init = encode_init,
.encode = encode_superframe, .encode2 = encode_superframe,
.close = ff_wma_end, .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"), .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
@ -423,7 +437,7 @@ AVCodec ff_wmav2_encoder = {
.id = CODEC_ID_WMAV2, .id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext), .priv_data_size = sizeof(WMACodecContext),
.init = encode_init, .init = encode_init,
.encode = encode_superframe, .encode2 = encode_superframe,
.close = ff_wma_end, .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"), .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),