X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fvorbisenc.c;h=35bdd57fb1a9cac463aa2c5136c38baac9c3ccf8;hb=bfe1cd80ebeab58cbc1c91ac766a96fce8e4ec1e;hp=5082efca766ffd35966a209b456ba0dccd8999d0;hpb=ffae713a5b3a0d20ff958d8bd58a052b495c38fd;p=ffmpeg

diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index 5082efca766..35bdd57fb1a 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -25,9 +25,11 @@
  */
 
 #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"
 
@@ -37,7 +39,7 @@
 #undef NDEBUG
 #include <assert.h>
 
-typedef struct {
+typedef struct vorbis_enc_codebook {
     int nentries;
     uint8_t *lens;
     uint32_t *codewords;
@@ -51,14 +53,14 @@ typedef struct {
     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;
@@ -69,7 +71,7 @@ typedef struct {
     vorbis_floor1_entry *list;
 } vorbis_enc_floor;
 
-typedef struct {
+typedef struct vorbis_enc_residue {
     int type;
     int begin;
     int end;
@@ -80,7 +82,7 @@ typedef struct {
     float (*maxes)[2];
 } vorbis_enc_residue;
 
-typedef struct {
+typedef struct vorbis_enc_mapping {
     int submaps;
     int *mux;
     int *floor;
@@ -90,12 +92,12 @@ typedef struct {
     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];
@@ -123,7 +125,7 @@ typedef struct {
     int nmodes;
     vorbis_enc_mode *modes;
 
-    int64_t sample_count;
+    int64_t next_pts;
 } vorbis_enc_context;
 
 #define MAX_CHANNELS     2
@@ -189,7 +191,7 @@ static int ready_codebook(vorbis_enc_codebook *cb)
                 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;
@@ -235,15 +237,15 @@ static int ready_residue(vorbis_enc_residue *rc, vorbis_enc_context *venc)
 }
 
 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);
@@ -339,7 +341,8 @@ static int create_vorbis_context(vorbis_enc_context *venc,
         };
         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);
@@ -727,7 +730,7 @@ static void floor_fit(vorbis_enc_context *venc, vorbis_enc_floor *fc,
 {
     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);
@@ -880,7 +883,7 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
     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]));
@@ -961,13 +964,13 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
     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)
@@ -986,9 +989,8 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
     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++)
@@ -1003,9 +1005,8 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
     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 {
@@ -1015,26 +1016,30 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
 }
 
 
-static int vorbis_encode_frame(AVCodecContext *avccontext,
-                               unsigned char *packets,
-                               int buf_size, void *data)
+static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                               const AVFrame *frame, int *got_packet_ptr)
 {
-    vorbis_enc_context *venc = avccontext->priv_data;
-    const signed short *audio = data;
-    int samples = data ? avccontext->frame_size : 0;
+    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;
     PutBitContext pb;
-    int i;
+    int i, ret;
 
     if (!apply_window_and_mdct(venc, audio, samples))
         return 0;
     samples = 1 << (venc->log2_blocksize[0] - 1);
 
-    init_put_bits(&pb, packets, buf_size);
+    if ((ret = ff_alloc_packet(avpkt, 8192))) {
+        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);
     }
 
@@ -1054,7 +1059,7 @@ static int vorbis_encode_frame(AVCodecContext *avccontext,
         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);
         }
     }
@@ -1078,20 +1083,31 @@ static int vorbis_encode_frame(AVCodecContext *avccontext,
 
     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);
     }
 
-    avccontext->coded_frame->pts = venc->sample_count;
-    venc->sample_count += avccontext->frame_size;
     flush_put_bits(&pb);
-    return put_bits_count(&pb) >> 3;
+    avpkt->size = put_bits_count(&pb) >> 3;
+
+    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(avctx, frame->pts);
+    } else {
+        avpkt->pts = venc->next_pts;
+    }
+    if (avpkt->pts != AV_NOPTS_VALUE)
+        venc->next_pts = avpkt->pts + avpkt->duration;
+
+    *got_packet_ptr = 1;
+    return 0;
 }
 
 
-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)
@@ -1143,58 +1159,53 @@ static av_cold int vorbis_encode_close(AVCodecContext *avccontext)
     ff_mdct_end(&venc->mdct[0]);
     ff_mdct_end(&venc->mdct[1]);
 
-    av_freep(&avccontext->coded_frame);
-    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);
-
-    avccontext->coded_frame = avcodec_alloc_frame();
-    if (!avccontext->coded_frame) {
-        ret = AVERROR(ENOMEM);
-        goto error;
-    }
+    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,
-    .encode         = vorbis_encode_frame,
+    .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 },
 };