* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+#include <float.h>
+
+#include "encode.h"
#include "opusenc.h"
#include "opus_pvq.h"
#include "opusenc_psy.h"
#include "opustab.h"
+#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
+#include "libavutil/mem_internal.h"
#include "libavutil/opt.h"
-#include "internal.h"
#include "bytestream.h"
#include "audio_frame_queue.h"
+#include "codec_internal.h"
typedef struct OpusEncContext {
AVClass *av_class;
AVCodecContext *avctx;
AudioFrameQueue afq;
AVFloatDSPContext *dsp;
- MDCT15Context *mdct[CELT_BLOCK_NB];
+ AVTXContext *tx[CELT_BLOCK_NB];
+ av_tx_fn tx_fn[CELT_BLOCK_NB];
CeltPVQ *pvq;
struct FFBufQueue bufqueue;
bytestream_put_buffer(&bs, "OpusHead", 8);
bytestream_put_byte (&bs, 0x1);
- bytestream_put_byte (&bs, avctx->channels);
+ bytestream_put_byte (&bs, avctx->ch_layout.nb_channels);
bytestream_put_le16 (&bs, avctx->initial_padding);
bytestream_put_le32 (&bs, avctx->sample_rate);
bytestream_put_le16 (&bs, 0x0);
float *src2 = &b->samples[CELT_OVERLAP*t];
s->dsp->vector_fmul(win, src1, ff_celt_window, 128);
s->dsp->vector_fmul_reverse(&win[CELT_OVERLAP], src2,
- ff_celt_window - 8, 128);
+ ff_celt_window_padded, 128);
src1 = src2;
- s->mdct[0]->mdct(s->mdct[0], b->coeffs + t, win, f->blocks);
+ s->tx_fn[0](s->tx[0], b->coeffs + t, win, sizeof(float)*f->blocks);
}
}
} else {
/* Samples, windowed */
s->dsp->vector_fmul_reverse(temp, b->samples + rwin,
- ff_celt_window - 8, 128);
+ ff_celt_window_padded, 128);
memcpy(win + lap_dst + blk_len, temp, CELT_OVERLAP*sizeof(float));
- s->mdct[f->size]->mdct(s->mdct[f->size], b->coeffs, win, 1);
+ s->tx_fn[f->size](s->tx[f->size], b->coeffs, win, sizeof(float));
}
}
static AVFrame *spawn_empty_frame(OpusEncContext *s)
{
AVFrame *f = av_frame_alloc();
+ int ret;
if (!f)
return NULL;
f->format = s->avctx->sample_fmt;
f->nb_samples = s->avctx->frame_size;
- f->channel_layout = s->avctx->channel_layout;
+ ret = av_channel_layout_copy(&f->ch_layout, &s->avctx->ch_layout);
+ if (ret < 0) {
+ av_frame_free(&f);
+ return NULL;
+ }
if (av_frame_get_buffer(f, 4)) {
av_frame_free(&f);
return NULL;
ff_bufqueue_add(avctx, &s->bufqueue, av_frame_clone(frame));
} else {
ff_opus_psy_signal_eof(&s->psyctx);
- if (!s->afq.remaining_samples || !avctx->frame_number)
+ if (!s->afq.remaining_samples || !avctx->frame_num)
return 0; /* We've been flushed and there's nothing left to encode */
}
/* Worst case toc + the frame lengths if needed */
alloc_size += 2 + s->packet.frames*2;
- if ((ret = ff_alloc_packet2(avctx, avpkt, alloc_size, 0)) < 0)
+ if ((ret = ff_alloc_packet(avctx, avpkt, alloc_size)) < 0)
return ret;
/* Assemble packet */
opus_packet_assembler(s, avpkt);
/* Update the psychoacoustic system */
- ff_opus_psy_postencode_update(&s->psyctx, s->frame, s->rc);
+ ff_opus_psy_postencode_update(&s->psyctx, s->frame);
/* Remove samples from queue and skip if needed */
ff_af_queue_remove(&s->afq, s->packet.frames*frame_size, &avpkt->pts, &avpkt->duration);
OpusEncContext *s = avctx->priv_data;
for (int i = 0; i < CELT_BLOCK_NB; i++)
- ff_mdct15_uninit(&s->mdct[i]);
+ av_tx_uninit(&s->tx[i]);
ff_celt_pvq_uninit(&s->pvq);
av_freep(&s->dsp);
ff_af_queue_close(&s->afq);
ff_opus_psy_end(&s->psyctx);
ff_bufqueue_discard_all(&s->bufqueue);
- av_freep(&avctx->extradata);
return 0;
}
OpusEncContext *s = avctx->priv_data;
s->avctx = avctx;
- s->channels = avctx->channels;
+ s->channels = avctx->ch_layout.nb_channels;
/* Opus allows us to change the framesize on each packet (and each packet may
* have multiple frames in it) but we can't change the codec's frame size on
return AVERROR(ENOMEM);
/* I have no idea why a base scaling factor of 68 works, could be the twiddles */
- for (int i = 0; i < CELT_BLOCK_NB; i++)
- if ((ret = ff_mdct15_init(&s->mdct[i], 0, i + 3, 68 << (CELT_BLOCK_NB - 1 - i))))
+ for (int i = 0; i < CELT_BLOCK_NB; i++) {
+ const float scale = 68 << (CELT_BLOCK_NB - 1 - i);
+ if ((ret = av_tx_init(&s->tx[i], &s->tx_fn[i], AV_TX_FLOAT_MDCT, 0, 15 << (i + 3), &scale, 0)))
return AVERROR(ENOMEM);
+ }
/* Zero out previous energy (matters for inter first frame) */
for (int ch = 0; ch < s->channels; ch++)
.version = LIBAVUTIL_VERSION_INT,
};
-static const AVCodecDefault opusenc_defaults[] = {
+static const FFCodecDefault opusenc_defaults[] = {
{ "b", "0" },
{ "compression_level", "10" },
{ NULL },
};
-AVCodec ff_opus_encoder = {
- .name = "opus",
- .long_name = NULL_IF_CONFIG_SMALL("Opus"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_OPUS,
+const FFCodec ff_opus_encoder = {
+ .p.name = "opus",
+ CODEC_LONG_NAME("Opus"),
+ .p.type = AVMEDIA_TYPE_AUDIO,
+ .p.id = AV_CODEC_ID_OPUS,
+ .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
+ AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_EXPERIMENTAL,
.defaults = opusenc_defaults,
- .priv_class = &opusenc_class,
+ .p.priv_class = &opusenc_class,
.priv_data_size = sizeof(OpusEncContext),
.init = opus_encode_init,
- .encode2 = opus_encode_frame,
+ FF_CODEC_ENCODE_CB(opus_encode_frame),
.close = opus_encode_end,
- .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
- .capabilities = AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
- .supported_samplerates = (const int []){ 48000, 0 },
- .channel_layouts = (const uint64_t []){ AV_CH_LAYOUT_MONO,
- AV_CH_LAYOUT_STEREO, 0 },
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
+ .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
+ .p.supported_samplerates = (const int []){ 48000, 0 },
+ CODEC_OLD_CHANNEL_LAYOUTS(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO)
+ .p.ch_layouts = (const AVChannelLayout []){ AV_CHANNEL_LAYOUT_MONO,
+ AV_CHANNEL_LAYOUT_STEREO, { 0 } },
+ .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
projects / platform / upstream / ffmpeg.git / blobdiff