[FFmpeg-cvslog] libmp3lame: use AVCodec.encode2()
Justin Ruggles
git at videolan.org
Thu Mar 22 01:33:57 CET 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sun Feb 19 13:16:34 2012 -0500| [3d853d7ab317a96a49873f3b3c1848a46f47c7ec] | committer: Justin Ruggles
libmp3lame: use AVCodec.encode2()
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3d853d7ab317a96a49873f3b3c1848a46f47c7ec
---
libavcodec/Makefile | 3 +-
libavcodec/libmp3lame.c | 71 +++++++++++++++++++++++++++++++++--------------
2 files changed, 52 insertions(+), 22 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 65385aa..2fa3130 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -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
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index d9f8120..c7abb99 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -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,
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