[FFmpeg-cvslog] avcodec/vorbisenc: Use a bufqueue in encoding with smaller lengths
Tyler Jones
git at videolan.org
Mon Jun 5 19:09:06 EEST 2017
ffmpeg | branch: master | Tyler Jones <tdjones879 at gmail.com> | Tue May 30 11:28:16 2017 -0600| [29c13fed68ac710a2584d31258ff9e576ee560e3] | committer: Rostislav Pehlivanov
avcodec/vorbisenc: Use a bufqueue in encoding with smaller lengths
Switching the vorbis encoder to use a buffer queue for input frames allows
saving lookahead samples more easily and safely for psychoacoustic systems,
requiring less pointer arithmetic in the case of transient windows.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=29c13fed68ac710a2584d31258ff9e576ee560e3
---
libavcodec/vorbisenc.c | 120 +++++++++++++++++++++++++++++++++++++++++++------
1 file changed, 106 insertions(+), 14 deletions(-)
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index f196d79d84..489feded27 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -1044,20 +1044,117 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
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;
+}
+
+static float **alloc_audio_arrays(int channels, int frame_size)
+{
+ float **audio = av_mallocz_array(channels, sizeof(float *));
+ if (!audio)
+ return NULL;
+
+ for (int ch = 0; ch < channels; ch++) {
+ audio[ch] = av_mallocz_array(frame_size, sizeof(float));
+ if (!audio[ch]) {
+ // alloc has failed, free everything allocated thus far
+ for (ch--; ch >= 0; ch--)
+ av_free(audio[ch]);
+ av_free(audio);
+ return NULL;
+ }
+ }
+
+ return audio;
+}
+
+/* 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][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;
+ float **audio = NULL;
+ 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 (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;
+
+ audio = alloc_audio_arrays(venc->channels, frame_size);
+ if (!audio)
+ return AVERROR(ENOMEM);
+
+ /* 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, audio, &samples, avctx->frame_size);
if (!apply_window_and_mdct(venc, audio, samples))
return 0;
- samples = 1 << (venc->log2_blocksize[0] - 1);
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
return ret;
@@ -1116,16 +1213,11 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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;
- }
- if (avpkt->pts != AV_NOPTS_VALUE)
- venc->next_pts = avpkt->pts + avpkt->duration;
+ for (int ch = 0; ch < venc->channels; ch++)
+ av_free(audio[ch]);
+ av_free(audio);
+ ff_af_queue_remove(&venc->afq, frame_size, &avpkt->pts, &avpkt->duration);
*got_packet_ptr = 1;
return 0;
}
@@ -1217,7 +1309,7 @@ static av_cold int vorbis_encode_init(AVCodecContext *avctx)
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);
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