*/
#include <float.h>
+
#include "avcodec.h"
-#include "dsputil.h"
#include "internal.h"
#include "fft.h"
+#include "mathops.h"
#include "vorbis.h"
#include "vorbis_enc_data.h"
#undef NDEBUG
#include <assert.h>
-typedef struct {
+typedef struct vorbis_enc_codebook {
int nentries;
uint8_t *lens;
uint32_t *codewords;
float *pow2;
} vorbis_enc_codebook;
-typedef struct {
+typedef struct vorbis_enc_floor_class {
int dim;
int subclass;
int masterbook;
int *books;
} vorbis_enc_floor_class;
-typedef struct {
+typedef struct vorbis_enc_floor {
int partitions;
int *partition_to_class;
int nclasses;
vorbis_floor1_entry *list;
} vorbis_enc_floor;
-typedef struct {
+typedef struct vorbis_enc_residue {
int type;
int begin;
int end;
float (*maxes)[2];
} vorbis_enc_residue;
-typedef struct {
+typedef struct vorbis_enc_mapping {
int submaps;
int *mux;
int *floor;
int *angle;
} vorbis_enc_mapping;
-typedef struct {
+typedef struct vorbis_enc_mode {
int blockflag;
int mapping;
} vorbis_enc_mode;
-typedef struct {
+typedef struct vorbis_enc_context {
int channels;
int sample_rate;
int log2_blocksize[2];
cb->pow2[i] += cb->dimensions[i * cb->ndimensions + j] * cb->dimensions[i * cb->ndimensions + j];
div *= vals;
}
- cb->pow2[i] /= 2.;
+ cb->pow2[i] /= 2.0;
}
}
return 0;
}
static int create_vorbis_context(vorbis_enc_context *venc,
- AVCodecContext *avccontext)
+ AVCodecContext *avctx)
{
vorbis_enc_floor *fc;
vorbis_enc_residue *rc;
vorbis_enc_mapping *mc;
int i, book, ret;
- venc->channels = avccontext->channels;
- venc->sample_rate = avccontext->sample_rate;
+ venc->channels = avctx->channels;
+ venc->sample_rate = avctx->sample_rate;
venc->log2_blocksize[0] = venc->log2_blocksize[1] = 11;
venc->ncodebooks = FF_ARRAY_ELEMS(cvectors);
};
fc->list[i].x = a[i - 2];
}
- ff_vorbis_ready_floor1_list(fc->list, fc->values);
+ if (ff_vorbis_ready_floor1_list(avctx, fc->list, fc->values))
+ return AVERROR_BUG;
venc->nresidues = 1;
venc->residues = av_malloc(sizeof(vorbis_enc_residue) * venc->nresidues);
{
int range = 255 / fc->multiplier + 1;
int i;
- float tot_average = 0.;
+ float tot_average = 0.0;
float averages[MAX_FLOOR_VALUES];
for (i = 0; i < fc->values; i++) {
averages[i] = get_floor_average(fc, coeffs, i);
assert(rc->type == 2);
assert(real_ch == 2);
for (p = 0; p < partitions; p++) {
- float max1 = 0., max2 = 0.;
+ float max1 = 0.0, max2 = 0.0;
int s = rc->begin + p * psize;
for (k = s; k < s + psize; k += 2) {
max1 = FFMAX(max1, fabs(coeffs[ k / real_ch]));
return 0;
}
-static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *audio,
- int samples)
+static int apply_window_and_mdct(vorbis_enc_context *venc,
+ float **audio, int samples)
{
- int i, j, channel;
+ int i, channel;
const float * win = venc->win[0];
int window_len = 1 << (venc->log2_blocksize[0] - 1);
- float n = (float)(1 << venc->log2_blocksize[0]) / 4.;
+ float n = (float)(1 << venc->log2_blocksize[0]) / 4.0;
// FIXME use dsp
if (!venc->have_saved && !samples)
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->samples + channel*window_len*2 + window_len;
- j = channel;
- for (i = 0; i < samples; i++, j += venc->channels)
- offset[i] = audio[j] / 32768. / n * win[window_len - i - 1];
+ for (i = 0; i < samples; i++)
+ offset[i] = audio[channel][i] / n * win[window_len - i - 1];
}
} else {
for (channel = 0; channel < venc->channels; channel++)
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->saved + channel * window_len;
- j = channel;
- for (i = 0; i < samples; i++, j += venc->channels)
- offset[i] = audio[j] / 32768. / n * win[i];
+ for (i = 0; i < samples; i++)
+ offset[i] = audio[channel][i] / n * win[i];
}
venc->have_saved = 1;
} else {
}
-static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
+static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
- vorbis_enc_context *venc = avccontext->priv_data;
- const int16_t *audio = frame ? (const int16_t *)frame->data[0] : NULL;
+ vorbis_enc_context *venc = avctx->priv_data;
+ float **audio = frame ? (float **)frame->extended_data : NULL;
int samples = frame ? frame->nb_samples : 0;
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
samples = 1 << (venc->log2_blocksize[0] - 1);
if ((ret = ff_alloc_packet(avpkt, 8192))) {
- av_log(avccontext, AV_LOG_ERROR, "Error getting output packet\n");
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
init_put_bits(&pb, avpkt->data, avpkt->size);
if (pb.size_in_bits - put_bits_count(&pb) < 1 + ilog(venc->nmodes - 1)) {
- av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
uint16_t posts[MAX_FLOOR_VALUES];
floor_fit(venc, fc, &venc->coeffs[i * samples], posts, samples);
if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * samples], samples)) {
- av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
}
if (residue_encode(venc, &venc->residues[mapping->residue[mapping->mux[0]]],
&pb, venc->coeffs, samples, venc->channels)) {
- av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+ av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
flush_put_bits(&pb);
avpkt->size = put_bits_count(&pb) >> 3;
- avpkt->duration = ff_samples_to_time_base(avccontext, avccontext->frame_size);
- if (frame)
+ avpkt->duration = ff_samples_to_time_base(avctx, avctx->frame_size);
+ if (frame) {
if (frame->pts != AV_NOPTS_VALUE)
- avpkt->pts = ff_samples_to_time_base(avccontext, frame->pts);
- else
+ avpkt->pts = ff_samples_to_time_base(avctx, frame->pts);
+ } else {
avpkt->pts = venc->next_pts;
+ }
if (avpkt->pts != AV_NOPTS_VALUE)
venc->next_pts = avpkt->pts + avpkt->duration;
}
-static av_cold int vorbis_encode_close(AVCodecContext *avccontext)
+static av_cold int vorbis_encode_close(AVCodecContext *avctx)
{
- vorbis_enc_context *venc = avccontext->priv_data;
+ vorbis_enc_context *venc = avctx->priv_data;
int i;
if (venc->codebooks)
ff_mdct_end(&venc->mdct[0]);
ff_mdct_end(&venc->mdct[1]);
-#if FF_API_OLD_ENCODE_AUDIO
- av_freep(&avccontext->coded_frame);
-#endif
- av_freep(&avccontext->extradata);
+ av_freep(&avctx->extradata);
return 0 ;
}
-static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
+static av_cold int vorbis_encode_init(AVCodecContext *avctx)
{
- vorbis_enc_context *venc = avccontext->priv_data;
+ vorbis_enc_context *venc = avctx->priv_data;
int ret;
- if (avccontext->channels != 2) {
- av_log(avccontext, AV_LOG_ERROR, "Current Libav Vorbis encoder only supports 2 channels.\n");
+ if (avctx->channels != 2) {
+ av_log(avctx, AV_LOG_ERROR, "Current Libav Vorbis encoder only supports 2 channels.\n");
return -1;
}
- if ((ret = create_vorbis_context(venc, avccontext)) < 0)
+ if ((ret = create_vorbis_context(venc, avctx)) < 0)
goto error;
- if (avccontext->flags & CODEC_FLAG_QSCALE)
- venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA / 10.;
+ avctx->bit_rate = 0;
+ if (avctx->flags & AV_CODEC_FLAG_QSCALE)
+ venc->quality = avctx->global_quality / (float)FF_QP2LAMBDA;
else
- venc->quality = 0.03;
+ venc->quality = 3.0;
venc->quality *= venc->quality;
- if ((ret = put_main_header(venc, (uint8_t**)&avccontext->extradata)) < 0)
+ if ((ret = put_main_header(venc, (uint8_t**)&avctx->extradata)) < 0)
goto error;
- avccontext->extradata_size = ret;
+ avctx->extradata_size = ret;
- avccontext->frame_size = 1 << (venc->log2_blocksize[0] - 1);
-
-#if FF_API_OLD_ENCODE_AUDIO
- avccontext->coded_frame = avcodec_alloc_frame();
- if (!avccontext->coded_frame) {
- ret = AVERROR(ENOMEM);
- goto error;
- }
-#endif
+ avctx->frame_size = 1 << (venc->log2_blocksize[0] - 1);
return 0;
error:
- vorbis_encode_close(avccontext);
+ vorbis_encode_close(avctx);
return ret;
}
AVCodec ff_vorbis_encoder = {
.name = "vorbis",
+ .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_VORBIS,
+ .id = AV_CODEC_ID_VORBIS,
.priv_data_size = sizeof(vorbis_enc_context),
.init = vorbis_encode_init,
.encode2 = vorbis_encode_frame,
.close = vorbis_encode_close,
- .capabilities= CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
- .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
+ .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_EXPERIMENTAL,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff