[FFmpeg-devel] [PATCH 2/2] avcodec/vorbisenc: Implement transient detection in Vorbis encoder

[Tyler Jones]

The existing AAC psychoacoustic system is used to detect transients within the
vorbis encoder. This is useful, in general, as an initial step in later 
utilizing
a complex psychoacoustic model for the vorbis encoder, but more specifically
allows the cacellation of pre-echo effects that frequently occur with this
codec.

Signed-off-by: Tyler Jones <tdjones...@gmail.com>
---
 libavcodec/psymodel.c  |  1 +
 libavcodec/vorbisenc.c | 60 ++++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 61 insertions(+)

diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index 2b5f111..38831ce 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -62,6 +62,7 @@ av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext 
*avctx, int num_lens,
 
     switch (ctx->avctx->codec_id) {
     case AV_CODEC_ID_AAC:
+    case AV_CODEC_ID_VORBIS:
         ctx->model = &ff_aac_psy_model;
         break;
     }
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index 2974ca2..e4ec822 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -33,6 +33,8 @@
 #include "vorbis.h"
 #include "vorbis_enc_data.h"
 
+#include "psymodel.h"
+
 #define BITSTREAM_WRITER_LE
 #include "put_bits.h"
 
@@ -126,6 +128,9 @@ typedef struct vorbis_enc_context {
     vorbis_enc_mode *modes;
 
     int64_t next_pts;
+
+    FFPsyContext psy;
+    struct FFPsyPreprocessContext* psypp;
 } vorbis_enc_context;
 
 #define MAX_CHANNELS     2
@@ -1024,10 +1029,38 @@ static int vorbis_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
     vorbis_enc_context *venc = avctx->priv_data;
     float **audio = frame ? (float **)frame->extended_data : NULL;
     int samples = frame ? frame->nb_samples : 0;
+    float *samples2, *la, *overlap;
     vorbis_enc_mode *mode;
     vorbis_enc_mapping *mapping;
     PutBitContext pb;
     int i, ret;
+    int start_ch, ch, chans, cur_channel;
+    FFPsyWindowInfo windows[MAX_CHANNELS];
+    enum WindowSequence window_sequence[MAX_CHANNELS];
+
+    if (!avctx->frame_number)
+        return 0;
+
+    if (venc->psypp)
+        ff_psy_preprocess(venc->psypp, audio, venc->channels);
+
+    if (frame) {
+        start_ch = 0;
+        cur_channel = 0;
+        for (i = 0; i < venc->channels - 1; i++) {
+            FFPsyWindowInfo* wi = windows + start_ch;
+            chans = 2;
+            for (ch = 0; ch < 2; ch++) {
+                cur_channel = start_ch + ch;
+                overlap = &audio[cur_channel][0];
+                samples2 = overlap + 1024;
+                la = samples2 + (448+64);
+                wi[ch] = venc->psy.model->window(&venc->psy, samples2, la,
+                                                 cur_channel, 
window_sequence[0]);
+            }
+            start_ch += chans;
+        }
+    }
 
     if (!apply_window_and_mdct(venc, audio, samples))
         return 0;
@@ -1158,7 +1191,10 @@ static av_cold int vorbis_encode_close(AVCodecContext 
*avctx)
 
     ff_mdct_end(&venc->mdct[0]);
     ff_mdct_end(&venc->mdct[1]);
+    ff_psy_end(&venc->psy);
 
+    if (venc->psypp)
+        ff_psy_preprocess_end(venc->psypp);
     av_freep(&avctx->extradata);
 
     return 0 ;
@@ -1168,6 +1204,10 @@ static av_cold int vorbis_encode_init(AVCodecContext 
*avctx)
 {
     vorbis_enc_context *venc = avctx->priv_data;
     int ret;
+    const uint8_t *sizes[MAX_CHANNELS];
+    uint8_t grouping[MAX_CHANNELS];
+    int lengths[MAX_CHANNELS];
+    int samplerate_index;
 
     if (avctx->channels != 2) {
         av_log(avctx, AV_LOG_ERROR, "Current FFmpeg Vorbis encoder only 
supports 2 channels.\n");
@@ -1190,6 +1230,26 @@ static av_cold int vorbis_encode_init(AVCodecContext 
*avctx)
 
     avctx->frame_size = 1 << (venc->log2_blocksize[0] - 1);
 
+    for (samplerate_index = 0; samplerate_index < 16; samplerate_index++)
+        if (avctx->sample_rate == mpeg4audio_sample_rates[samplerate_index])
+            break;
+    if (samplerate_index == 16 ||
+        samplerate_index >= ff_vorbis_swb_size_1024_len ||
+        samplerate_index >= ff_vorbis_swb_size_128_len)
+        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", 
avctx->sample_rate);
+
+    sizes[0]   = ff_vorbis_swb_size_1024[samplerate_index];
+    sizes[1]   = ff_vorbis_swb_size_128[samplerate_index];
+    lengths[0] = ff_vorbis_num_swb_1024[samplerate_index];
+    lengths[1] = ff_vorbis_num_swb_128[samplerate_index];
+    grouping[0] = 1;
+
+    if ((ret = ff_psy_init(&venc->psy, avctx, 2,
+                           sizes, lengths,
+                           1, grouping)) < 0)
+        goto error;
+    venc->psypp = ff_psy_preprocess_init(avctx);
+
     return 0;
 error:
     vorbis_encode_close(avctx);
-- 
2.7.4

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