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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

g722enc: use AVCodec.encode2()

FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
This commit is contained in:
Justin Ruggles 2012-02-04 11:26:33 -05:00
parent 910bdb9a42
commit bb03b6f7b1
2 changed files with 41 additions and 19 deletions

View File

@ -28,6 +28,7 @@
*/
#include "avcodec.h"
#include "internal.h"
#include "g722.h"
#define FREEZE_INTERVAL 128
@ -50,6 +51,9 @@ static av_cold int g722_encode_close(AVCodecContext *avctx)
av_freep(&c->node_buf[i]);
av_freep(&c->nodep_buf[i]);
}
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
return 0;
}
@ -104,6 +108,7 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
a common packet size for VoIP applications */
avctx->frame_size = 320;
}
avctx->delay = 22;
if (avctx->trellis) {
/* validate trellis */
@ -116,6 +121,14 @@ static av_cold int g722_encode_init(AVCodecContext * avctx)
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
g722_encode_close(avctx);
@ -345,27 +358,36 @@ static void g722_encode_no_trellis(G722Context *c,
encode_byte(c, dst++, &samples[i]);
}
static int g722_encode_frame(AVCodecContext *avctx,
uint8_t *dst, int buf_size, void *data)
static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G722Context *c = avctx->priv_data;
const int16_t *samples = data;
int nb_samples;
const int16_t *samples = (const int16_t *)frame->data[0];
int nb_samples, out_size, ret;
nb_samples = avctx->frame_size - (avctx->frame_size & 1);
if (avctx->trellis)
g722_encode_trellis(c, avctx->trellis, dst, nb_samples, samples);
else
g722_encode_no_trellis(c, dst, nb_samples, samples);
/* handle last frame with odd frame_size */
if (nb_samples < avctx->frame_size) {
int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
encode_byte(c, &dst[nb_samples >> 1], last_samples);
out_size = (frame->nb_samples + 1) / 2;
if ((ret = ff_alloc_packet(avpkt, out_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
return (avctx->frame_size + 1) >> 1;
nb_samples = frame->nb_samples - (frame->nb_samples & 1);
if (avctx->trellis)
g722_encode_trellis(c, avctx->trellis, avpkt->data, nb_samples, samples);
else
g722_encode_no_trellis(c, avpkt->data, nb_samples, samples);
/* handle last frame with odd frame_size */
if (nb_samples < frame->nb_samples) {
int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
encode_byte(c, &avpkt->data[nb_samples >> 1], last_samples);
}
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_adpcm_g722_encoder = {
@ -375,7 +397,7 @@ AVCodec ff_adpcm_g722_encoder = {
.priv_data_size = sizeof(G722Context),
.init = g722_encode_init,
.close = g722_encode_close,
.encode = g722_encode_frame,
.encode2 = g722_encode_frame,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},

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@ -1,4 +1,4 @@
7b0492eee76b04b710990235f97a0bf2 *./tests/data/acodec/g722.wav
48053 ./tests/data/acodec/g722.wav
f30e8e99cfd3f38ba66f1d4131602a19 *./tests/data/acodec/g722.wav
48053 ./tests/data/acodec/g722.wav
b5568e0e3930ff563824156e8e1015f0 *./tests/data/g722.acodec.out.wav
stddev: 8939.44 PSNR: 17.30 MAXDIFF:40370 bytes: 191980/ 1058400