*/
#include <float.h>
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis_enc_data.h"
+#include "audio_frame_queue.h"
+#include "libavfilter/bufferqueue.h"
+
#define BITSTREAM_WRITER_LE
#include "put_bits.h"
float *samples;
float *floor; // also used for tmp values for mdct
float *coeffs; // also used for residue after floor
+ float *scratch; // used for tmp values for psy model
float quality;
+ AudioFrameQueue afq;
+ struct FFBufQueue bufqueue;
+
int ncodebooks;
vorbis_enc_codebook *codebooks;
vorbis_enc_mode *modes;
int64_t next_pts;
+
+ AVFloatDSPContext *fdsp;
} vorbis_enc_context;
#define MAX_CHANNELS 2
return 0;
}
+static av_cold int dsp_init(AVCodecContext *avctx, vorbis_enc_context *venc)
+{
+ int ret = 0;
+
+ venc->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+ if (!venc->fdsp)
+ return AVERROR(ENOMEM);
+
+ // init windows
+ venc->win[0] = ff_vorbis_vwin[venc->log2_blocksize[0] - 6];
+ venc->win[1] = ff_vorbis_vwin[venc->log2_blocksize[1] - 6];
+
+ if ((ret = ff_mdct_init(&venc->mdct[0], venc->log2_blocksize[0], 0, 1.0)) < 0)
+ return ret;
+ if ((ret = ff_mdct_init(&venc->mdct[1], venc->log2_blocksize[1], 0, 1.0)) < 0)
+ return ret;
+
+ return 0;
+}
+
static int create_vorbis_context(vorbis_enc_context *venc,
AVCodecContext *avctx)
{
venc->samples = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]));
venc->floor = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]) / 2);
venc->coeffs = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]) / 2);
- if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs)
- return AVERROR(ENOMEM);
+ venc->scratch = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]) / 2);
- venc->win[0] = ff_vorbis_vwin[venc->log2_blocksize[0] - 6];
- venc->win[1] = ff_vorbis_vwin[venc->log2_blocksize[1] - 6];
+ if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs || !venc->scratch)
+ return AVERROR(ENOMEM);
- if ((ret = ff_mdct_init(&venc->mdct[0], venc->log2_blocksize[0], 0, 1.0)) < 0)
- return ret;
- if ((ret = ff_mdct_init(&venc->mdct[1], venc->log2_blocksize[1], 0, 1.0)) < 0)
+ if ((ret = dsp_init(avctx, venc)) < 0)
return ret;
return 0;
}
static int apply_window_and_mdct(vorbis_enc_context *venc,
- float **audio, int samples)
+ float *audio, int samples)
{
- int i, channel;
+ int channel;
const float * win = venc->win[0];
int window_len = 1 << (venc->log2_blocksize[0] - 1);
float n = (float)(1 << venc->log2_blocksize[0]) / 4.0;
- // FIXME use dsp
+ AVFloatDSPContext *fdsp = venc->fdsp;
if (!venc->have_saved && !samples)
return 0;
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
- float * offset = venc->samples + channel*window_len*2 + window_len;
- for (i = 0; i < samples; i++)
- offset[i] = audio[channel][i] / n * win[window_len - i - 1];
+ float *offset = venc->samples + channel * window_len * 2 + window_len;
+
+ fdsp->vector_fmul_reverse(offset, audio + channel * window_len, win, samples);
+ fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
}
} else {
for (channel = 0; channel < venc->channels; channel++)
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->saved + channel * window_len;
- for (i = 0; i < samples; i++)
- offset[i] = audio[channel][i] / n * win[i];
+
+ fdsp->vector_fmul(offset, audio + channel * window_len, win, samples);
+ fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
}
venc->have_saved = 1;
} else {
return 1;
}
+/* Used for padding the last encoded packet */
+static AVFrame *spawn_empty_frame(AVCodecContext *avctx, int channels)
+{
+ AVFrame *f = av_frame_alloc();
+
+ if (!f)
+ return NULL;
+
+ f->format = avctx->sample_fmt;
+ f->nb_samples = avctx->frame_size;
+ f->channel_layout = avctx->channel_layout;
+
+ if (av_frame_get_buffer(f, 4)) {
+ av_frame_free(&f);
+ return NULL;
+ }
+
+ for (int ch = 0; ch < channels; ch++) {
+ size_t bps = av_get_bytes_per_sample(f->format);
+ memset(f->extended_data[ch], 0, bps * f->nb_samples);
+ }
+ return f;
+}
+
+/* Concatenate audio frames into an appropriately sized array of samples */
+static void move_audio(vorbis_enc_context *venc, float *audio, int *samples, int sf_size)
+{
+ AVFrame *cur = NULL;
+ int frame_size = 1 << (venc->log2_blocksize[1] - 1);
+ int subframes = frame_size / sf_size;
+
+ for (int sf = 0; sf < subframes; sf++) {
+ cur = ff_bufqueue_get(&venc->bufqueue);
+ *samples += cur->nb_samples;
+
+ for (int ch = 0; ch < venc->channels; ch++) {
+ const float *input = (float *) cur->extended_data[ch];
+ const size_t len = cur->nb_samples * sizeof(float);
+ memcpy(audio + ch*frame_size + sf*sf_size, input, len);
+ }
+ av_frame_free(&cur);
+ }
+}
+
static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
vorbis_enc_context *venc = avctx->priv_data;
- float **audio = frame ? (float **)frame->extended_data : NULL;
- int samples = frame ? frame->nb_samples : 0;
+ int i, ret, need_more;
+ int samples = 0, frame_size = 1 << (venc->log2_blocksize[1] - 1);
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
PutBitContext pb;
- int i, ret;
- if (!apply_window_and_mdct(venc, audio, samples))
+ if (frame) {
+ if ((ret = ff_af_queue_add(&venc->afq, frame)) < 0)
+ return ret;
+ ff_bufqueue_add(avctx, &venc->bufqueue, av_frame_clone(frame));
+ } else
+ if (!venc->afq.remaining_samples)
+ return 0;
+
+ need_more = venc->bufqueue.available * avctx->frame_size < frame_size;
+ need_more = frame && need_more;
+ if (need_more)
+ return 0;
+
+ /* Pad the bufqueue with empty frames for encoding the last packet. */
+ if (!frame) {
+ if (venc->bufqueue.available * avctx->frame_size < frame_size) {
+ int frames_needed = (frame_size/avctx->frame_size) - venc->bufqueue.available;
+
+ for (int i = 0; i < frames_needed; i++) {
+ AVFrame *empty = spawn_empty_frame(avctx, venc->channels);
+ if (!empty)
+ return AVERROR(ENOMEM);
+
+ ff_bufqueue_add(avctx, &venc->bufqueue, empty);
+ }
+ }
+ }
+
+ move_audio(venc, venc->scratch, &samples, avctx->frame_size);
+
+ if (!apply_window_and_mdct(venc, venc->scratch, samples))
return 0;
- samples = 1 << (venc->log2_blocksize[0] - 1);
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
return ret;
flush_put_bits(&pb);
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;
+ ff_af_queue_remove(&venc->afq, frame_size, &avpkt->pts, &avpkt->duration);
+
+ if (frame_size > avpkt->duration) {
+ uint8_t *side = av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
+ if (!side)
+ return AVERROR(ENOMEM);
+ AV_WL32(&side[4], frame_size - avpkt->duration);
}
- if (avpkt->pts != AV_NOPTS_VALUE)
- venc->next_pts = avpkt->pts + avpkt->duration;
*got_packet_ptr = 1;
return 0;
av_freep(&venc->samples);
av_freep(&venc->floor);
av_freep(&venc->coeffs);
+ av_freep(&venc->scratch);
+ av_freep(&venc->fdsp);
ff_mdct_end(&venc->mdct[0]);
ff_mdct_end(&venc->mdct[1]);
+ ff_af_queue_close(&venc->afq);
+ ff_bufqueue_discard_all(&venc->bufqueue);
av_freep(&avctx->extradata);
goto error;
avctx->extradata_size = ret;
- avctx->frame_size = 1 << (venc->log2_blocksize[0] - 1);
+ avctx->frame_size = 64;
+
+ ff_af_queue_init(avctx, &venc->afq);
return 0;
error:
projects / ffmpeg / blobdiff