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libmp3lame: use AVCodec.encode2()

This commit is contained in:
Justin Ruggles 2012-02-19 13:16:34 -05:00
parent 1987a940b7
commit 3d853d7ab3
2 changed files with 52 additions and 22 deletions
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3
libavcodec/Makefile
View File

@ -586,7 +586,8 @@ OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o \
audio_frame_queue.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o

71
libavcodec/libmp3lame.c
View File

@ -30,6 +30,7 @@
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
@ -44,6 +45,7 @@ typedef struct LAMEContext {
int buffer_index;
int reservoir;
void *planar_samples[2];
AudioFrameQueue afq;
} LAMEContext;
@ -51,10 +53,14 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{
LAMEContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->planar_samples[0]);
av_freep(&s->planar_samples[1]);
ff_af_queue_close(&s->afq);
lame_close(s->gfp);
return 0;
}
@ -109,12 +115,19 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
goto error;
}
/* get encoder delay */
avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
ff_af_queue_init(avctx, &s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
/* sample format */
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
@ -142,67 +155,67 @@ error:
const type *input = samples; \
type *output = s->planar_samples[ch]; \
input += ch; \
for (i = 0; i < s->avctx->frame_size; i++) { \
for (i = 0; i < nb_samples; i++) { \
output[i] = *input * scale; \
input += s->avctx->channels; \
} \
} \
} while (0)
static int encode_frame_int16(LAMEContext *s, void *samples)
static int encode_frame_int16(LAMEContext *s, void *samples, int nb_samples)
{
if (s->avctx->channels > 1) {
return lame_encode_buffer_interleaved(s->gfp, samples,
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
return lame_encode_buffer(s->gfp, samples, NULL, s->avctx->frame_size,
return lame_encode_buffer(s->gfp, samples, NULL, nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
}
static int encode_frame_int32(LAMEContext *s, void *samples)
static int encode_frame_int32(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(int32_t, 1);
return lame_encode_buffer_int(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int encode_frame_float(LAMEContext *s, void *samples)
static int encode_frame_float(LAMEContext *s, void *samples, int nb_samples)
{
DEINTERLEAVE(float, 32768.0f);
return lame_encode_buffer_float(s->gfp,
s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size,
nb_samples,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
}
static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len;
int len, ret;
int lame_result;
if (data) {
if (frame) {
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16:
lame_result = encode_frame_int16(s, data);
lame_result = encode_frame_int16(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_S32:
lame_result = encode_frame_int32(s, data);
lame_result = encode_frame_int32(s, frame->data[0], frame->nb_samples);
break;
case AV_SAMPLE_FMT_FLT:
lame_result = encode_frame_float(s, data);
lame_result = encode_frame_float(s, frame->data[0], frame->nb_samples);
break;
default:
return AVERROR_BUG;
@ -221,6 +234,12 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
}
s->buffer_index += lame_result;
/* add current frame to the queue */
if (frame) {
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
}
/* Move 1 frame from the LAME buffer to the output packet, if available.
We have to parse the first frame header in the output buffer to
determine the frame size. */
@ -234,12 +253,22 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index);
if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len);
if ((ret = ff_alloc_packet(avpkt, len))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
memcpy(avpkt->data, s->buffer, len);
s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index);
return len;
} else
return 0;
/* Get the next frame pts/duration */
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = len;
*got_packet_ptr = 1;
}
return 0;
}
#define OFFSET(x) offsetof(LAMEContext, x)
@ -271,9 +300,9 @@ AVCodec ff_libmp3lame_encoder = {
.id = CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init,
.encode = mp3lame_encode_frame,
.encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close,
.capabilities = CODEC_CAP_DELAY,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,