libfaac: use AVCodec.encode2()
Encoder output is delayed by several frames, so we keep a queue of input frame timing info to match up with corresponding output packets.
This commit is contained in:
parent
59041fd053
commit
d1afb2f94e
@ -581,7 +581,7 @@ OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
|
|||||||
|
|
||||||
# external codec libraries
|
# external codec libraries
|
||||||
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
|
OBJS-$(CONFIG_LIBDIRAC_DECODER) += libdiracdec.o
|
||||||
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
|
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o audio_frame_queue.o
|
||||||
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
|
OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
|
||||||
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
|
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
|
||||||
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
|
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
|
||||||
|
@ -24,11 +24,19 @@
|
|||||||
* Interface to libfaac for aac encoding.
|
* Interface to libfaac for aac encoding.
|
||||||
*/
|
*/
|
||||||
|
|
||||||
#include "avcodec.h"
|
|
||||||
#include <faac.h>
|
#include <faac.h>
|
||||||
|
|
||||||
|
#include "avcodec.h"
|
||||||
|
#include "audio_frame_queue.h"
|
||||||
|
#include "internal.h"
|
||||||
|
|
||||||
|
|
||||||
|
/* libfaac has an encoder delay of 1024 samples */
|
||||||
|
#define FAAC_DELAY_SAMPLES 1024
|
||||||
|
|
||||||
typedef struct FaacAudioContext {
|
typedef struct FaacAudioContext {
|
||||||
faacEncHandle faac_handle;
|
faacEncHandle faac_handle;
|
||||||
|
AudioFrameQueue afq;
|
||||||
} FaacAudioContext;
|
} FaacAudioContext;
|
||||||
|
|
||||||
|
|
||||||
@ -36,11 +44,15 @@ static av_cold int Faac_encode_close(AVCodecContext *avctx)
|
|||||||
{
|
{
|
||||||
FaacAudioContext *s = avctx->priv_data;
|
FaacAudioContext *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);
|
||||||
|
ff_af_queue_close(&s->afq);
|
||||||
|
|
||||||
if (s->faac_handle)
|
if (s->faac_handle)
|
||||||
faacEncClose(s->faac_handle);
|
faacEncClose(s->faac_handle);
|
||||||
|
|
||||||
return 0;
|
return 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
@ -109,11 +121,13 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
|
|||||||
|
|
||||||
avctx->frame_size = samples_input / avctx->channels;
|
avctx->frame_size = samples_input / avctx->channels;
|
||||||
|
|
||||||
|
#if FF_API_OLD_ENCODE_AUDIO
|
||||||
avctx->coded_frame= avcodec_alloc_frame();
|
avctx->coded_frame= avcodec_alloc_frame();
|
||||||
if (!avctx->coded_frame) {
|
if (!avctx->coded_frame) {
|
||||||
ret = AVERROR(ENOMEM);
|
ret = AVERROR(ENOMEM);
|
||||||
goto error;
|
goto error;
|
||||||
}
|
}
|
||||||
|
#endif
|
||||||
|
|
||||||
/* Set decoder specific info */
|
/* Set decoder specific info */
|
||||||
avctx->extradata_size = 0;
|
avctx->extradata_size = 0;
|
||||||
@ -144,26 +158,52 @@ static av_cold int Faac_encode_init(AVCodecContext *avctx)
|
|||||||
goto error;
|
goto error;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
avctx->delay = FAAC_DELAY_SAMPLES;
|
||||||
|
ff_af_queue_init(avctx, &s->afq);
|
||||||
|
|
||||||
return 0;
|
return 0;
|
||||||
error:
|
error:
|
||||||
Faac_encode_close(avctx);
|
Faac_encode_close(avctx);
|
||||||
return ret;
|
return ret;
|
||||||
}
|
}
|
||||||
|
|
||||||
static int Faac_encode_frame(AVCodecContext *avctx,
|
static int Faac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
||||||
unsigned char *frame, int buf_size, void *data)
|
const AVFrame *frame, int *got_packet_ptr)
|
||||||
{
|
{
|
||||||
FaacAudioContext *s = avctx->priv_data;
|
FaacAudioContext *s = avctx->priv_data;
|
||||||
int bytes_written;
|
int bytes_written, ret;
|
||||||
int num_samples = data ? avctx->frame_size : 0;
|
int num_samples = frame ? frame->nb_samples : 0;
|
||||||
|
void *samples = frame ? frame->data[0] : NULL;
|
||||||
|
|
||||||
bytes_written = faacEncEncode(s->faac_handle,
|
if ((ret = ff_alloc_packet(avpkt, (7 + 768) * avctx->channels))) {
|
||||||
data,
|
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
bytes_written = faacEncEncode(s->faac_handle, samples,
|
||||||
num_samples * avctx->channels,
|
num_samples * avctx->channels,
|
||||||
frame,
|
avpkt->data, avpkt->size);
|
||||||
buf_size);
|
if (bytes_written < 0) {
|
||||||
|
av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
|
||||||
|
return bytes_written;
|
||||||
|
}
|
||||||
|
|
||||||
return bytes_written;
|
/* add current frame to the queue */
|
||||||
|
if (frame) {
|
||||||
|
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (!bytes_written)
|
||||||
|
return 0;
|
||||||
|
|
||||||
|
/* Get the next frame pts/duration */
|
||||||
|
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
|
||||||
|
&avpkt->duration);
|
||||||
|
|
||||||
|
avpkt->size = bytes_written;
|
||||||
|
*got_packet_ptr = 1;
|
||||||
|
return 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
static const AVProfile profiles[] = {
|
static const AVProfile profiles[] = {
|
||||||
@ -180,7 +220,7 @@ AVCodec ff_libfaac_encoder = {
|
|||||||
.id = CODEC_ID_AAC,
|
.id = CODEC_ID_AAC,
|
||||||
.priv_data_size = sizeof(FaacAudioContext),
|
.priv_data_size = sizeof(FaacAudioContext),
|
||||||
.init = Faac_encode_init,
|
.init = Faac_encode_init,
|
||||||
.encode = Faac_encode_frame,
|
.encode2 = Faac_encode_frame,
|
||||||
.close = Faac_encode_close,
|
.close = Faac_encode_close,
|
||||||
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
|
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
|
||||||
.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},
|
||||||
|
Loading…
x
Reference in New Issue
Block a user