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libaacplus: switch to encode2()

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Paul B Mahol
2012-03-24 17:41:55 +00:00
committed by Michael Niedermayer
parent 15890d67c1
commit bcae7f432a
1 changed files with 24 additions and 16 deletions
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40
libavcodec/libaacplus.c
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@@ -24,18 +24,21 @@
* Interface to libaacplus for aac+ (sbr+ps) encoding. * Interface to libaacplus for aac+ (sbr+ps) encoding.
*/ */
#include "avcodec.h"
#include <aacplus.h> #include <aacplus.h>
#include "avcodec.h"
#include "internal.h"
typedef struct aacPlusAudioContext { typedef struct aacPlusAudioContext {
aacplusEncHandle aacplus_handle; aacplusEncHandle aacplus_handle;
unsigned long max_output_bytes;
unsigned long samples_input;
} aacPlusAudioContext; } aacPlusAudioContext;
static av_cold int aacPlus_encode_init(AVCodecContext *avctx) static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
{ {
aacPlusAudioContext *s = avctx->priv_data; aacPlusAudioContext *s = avctx->priv_data;
aacplusEncConfiguration *aacplus_cfg; aacplusEncConfiguration *aacplus_cfg;
unsigned long samples_input, max_bytes_output;
/* number of channels */ /* number of channels */
if (avctx->channels < 1 || avctx->channels > 2) { if (avctx->channels < 1 || avctx->channels > 2) {
@@ -43,9 +46,8 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
return -1; return -1;
} }
s->aacplus_handle = aacplusEncOpen(avctx->sample_rate, s->aacplus_handle = aacplusEncOpen(avctx->sample_rate, avctx->channels,
avctx->channels, &s->samples_input, &s->max_output_bytes);
&samples_input, &max_bytes_output);
if(!s->aacplus_handle) { if(!s->aacplus_handle) {
av_log(avctx, AV_LOG_ERROR, "can't open encoder\n"); av_log(avctx, AV_LOG_ERROR, "can't open encoder\n");
return -1; return -1;
@@ -70,10 +72,12 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
return -1; return -1;
} }
avctx->frame_size = samples_input / avctx->channels; avctx->frame_size = s->samples_input / avctx->channels;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1; avctx->coded_frame->key_frame= 1;
#endif
/* Set decoder specific info */ /* Set decoder specific info */
avctx->extradata_size = 0; avctx->extradata_size = 0;
@@ -95,26 +99,30 @@ static av_cold int aacPlus_encode_init(AVCodecContext *avctx)
return 0; return 0;
} }
static int aacPlus_encode_frame(AVCodecContext *avctx, static int aacPlus_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
unsigned char *frame, int buf_size, void *data) const AVFrame *frame, int *got_packet)
{ {
aacPlusAudioContext *s = avctx->priv_data; aacPlusAudioContext *s = avctx->priv_data;
int bytes_written; int32_t *input_buffer = (int32_t *)frame->data[0];
int ret;
bytes_written = aacplusEncEncode(s->aacplus_handle, if ((ret = ff_alloc_packet2(avctx, pkt, s->max_output_bytes)))
data, return ret;
avctx->frame_size * avctx->channels,
frame,
buf_size);
return bytes_written; pkt->size = aacplusEncEncode(s->aacplus_handle, input_buffer,
s->samples_input, pkt->data, pkt->size);
*got_packet = 1;
pkt->pts = frame->pts;
return 0;
} }
static av_cold int aacPlus_encode_close(AVCodecContext *avctx) static av_cold int aacPlus_encode_close(AVCodecContext *avctx)
{ {
aacPlusAudioContext *s = avctx->priv_data; aacPlusAudioContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame); av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata); av_freep(&avctx->extradata);
aacplusEncClose(s->aacplus_handle); aacplusEncClose(s->aacplus_handle);
@@ -127,7 +135,7 @@ AVCodec ff_libaacplus_encoder = {
.id = CODEC_ID_AAC, .id = CODEC_ID_AAC,
.priv_data_size = sizeof(aacPlusAudioContext), .priv_data_size = sizeof(aacPlusAudioContext),
.init = aacPlus_encode_init, .init = aacPlus_encode_init,
.encode = aacPlus_encode_frame, .encode2 = aacPlus_encode_frame,
.close = aacPlus_encode_close, .close = aacPlus_encode_close,
.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("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"), .long_name = NULL_IF_CONFIG_SMALL("libaacplus AAC+ (Advanced Audio Codec with SBR+PS)"),