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libmp3lame: use AVCodec.encode2()

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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
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@ -586,7 +586,8 @@ 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-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += 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-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += 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/log.h"
#include "libavutil/opt.h" #include "libavutil/opt.h"
#include "avcodec.h" #include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h" #include "internal.h"
#include "mpegaudio.h" #include "mpegaudio.h"
#include "mpegaudiodecheader.h" #include "mpegaudiodecheader.h"
@ -44,6 +45,7 @@ typedef struct LAMEContext {
int buffer_index; int buffer_index;
int reservoir; int reservoir;
void *planar_samples[2]; void *planar_samples[2];
AudioFrameQueue afq;
} LAMEContext; } LAMEContext;
@ -51,10 +53,14 @@ static av_cold int mp3lame_encode_close(AVCodecContext *avctx)
{ {
LAMEContext *s = avctx->priv_data; LAMEContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame); av_freep(&avctx->coded_frame);
#endif
av_freep(&s->planar_samples[0]); av_freep(&s->planar_samples[0]);
av_freep(&s->planar_samples[1]); av_freep(&s->planar_samples[1]);
ff_af_queue_close(&s->afq);
lame_close(s->gfp); lame_close(s->gfp);
return 0; return 0;
} }
@ -109,12 +115,19 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx)
goto error; 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); avctx->frame_size = lame_get_framesize(s->gfp);
#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
/* sample format */ /* sample format */
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 || if (avctx->sample_fmt == AV_SAMPLE_FMT_S32 ||
@ -142,67 +155,67 @@ error:
const type *input = samples; \ const type *input = samples; \
type *output = s->planar_samples[ch]; \ type *output = s->planar_samples[ch]; \
input += ch; \ input += ch; \
for (i = 0; i < s->avctx->frame_size; i++) { \ for (i = 0; i < nb_samples; i++) { \
output[i] = *input * scale; \ output[i] = *input * scale; \
input += s->avctx->channels; \ input += s->avctx->channels; \
} \ } \
} \ } \
} while (0) } 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) { if (s->avctx->channels > 1) {
return lame_encode_buffer_interleaved(s->gfp, samples, return lame_encode_buffer_interleaved(s->gfp, samples,
s->avctx->frame_size, nb_samples,
s->buffer + s->buffer_index, s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index); BUFFER_SIZE - s->buffer_index);
} else { } 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, s->buffer + s->buffer_index,
BUFFER_SIZE - 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); DEINTERLEAVE(int32_t, 1);
return lame_encode_buffer_int(s->gfp, return lame_encode_buffer_int(s->gfp,
s->planar_samples[0], s->planar_samples[1], s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size, nb_samples,
s->buffer + s->buffer_index, s->buffer + s->buffer_index,
BUFFER_SIZE - 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); DEINTERLEAVE(float, 32768.0f);
return lame_encode_buffer_float(s->gfp, return lame_encode_buffer_float(s->gfp,
s->planar_samples[0], s->planar_samples[1], s->planar_samples[0], s->planar_samples[1],
s->avctx->frame_size, nb_samples,
s->buffer + s->buffer_index, s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index); BUFFER_SIZE - s->buffer_index);
} }
static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame, static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
int buf_size, void *data) const AVFrame *frame, int *got_packet_ptr)
{ {
LAMEContext *s = avctx->priv_data; LAMEContext *s = avctx->priv_data;
MPADecodeHeader hdr; MPADecodeHeader hdr;
int len; int len, ret;
int lame_result; int lame_result;
if (data) { if (frame) {
switch (avctx->sample_fmt) { switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_S16: 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; break;
case AV_SAMPLE_FMT_S32: 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; break;
case AV_SAMPLE_FMT_FLT: 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; break;
default: default:
return AVERROR_BUG; return AVERROR_BUG;
@ -221,6 +234,12 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
} }
s->buffer_index += lame_result; 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. /* 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 We have to parse the first frame header in the output buffer to
determine the frame size. */ 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, av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index); s->buffer_index);
if (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; s->buffer_index -= len;
memmove(s->buffer, s->buffer + len, s->buffer_index); memmove(s->buffer, s->buffer + len, s->buffer_index);
return len;
} else /* Get the next frame pts/duration */
return 0; 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) #define OFFSET(x) offsetof(LAMEContext, x)
@ -271,9 +300,9 @@ AVCodec ff_libmp3lame_encoder = {
.id = CODEC_ID_MP3, .id = CODEC_ID_MP3,
.priv_data_size = sizeof(LAMEContext), .priv_data_size = sizeof(LAMEContext),
.init = mp3lame_encode_init, .init = mp3lame_encode_init,
.encode = mp3lame_encode_frame, .encode2 = mp3lame_encode_frame,
.close = mp3lame_encode_close, .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, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,