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vorbisenc: Separate copying audio samples from windowing

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Audio samples are shifted around when copying from the frame queue so that
analysis can be done without negatively impacting calculation of the MDCT.

Window coefficients are applied to the current two overlapped windows
simultaneously instead of applying overlap for the next frame ahead of time.
This improves readability when applying windows of varying lengths.

Signed-off-by: Tyler Jones <tdjones879@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit is contained in:
Tyler Jones
2017-06-14 14:58:22 -06:00
committed by Rostislav Pehlivanov
parent 9b667f609c
commit 5a2ad7ede3
1 changed files with 32 additions and 44 deletions
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76
libavcodec/vorbisenc.c
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@@ -453,7 +453,7 @@ static int create_vorbis_context(vorbis_enc_context *venc,
venc->samples = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1])); 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->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); venc->coeffs = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]) / 2);
venc->scratch = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]) / 2); venc->scratch = av_malloc_array(sizeof(float) * venc->channels, (1 << venc->log2_blocksize[1]));
if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs || !venc->scratch) if (!venc->saved || !venc->samples || !venc->floor || !venc->coeffs || !venc->scratch)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);
@@ -994,8 +994,7 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
return 0; return 0;
} }
static int apply_window_and_mdct(vorbis_enc_context *venc, static int apply_window_and_mdct(vorbis_enc_context *venc, int samples)
float *audio, int samples)
{ {
int channel; int channel;
const float * win = venc->win[0]; const float * win = venc->win[0];
@@ -1003,46 +1002,19 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
float n = (float)(1 << venc->log2_blocksize[0]) / 4.0; float n = (float)(1 << venc->log2_blocksize[0]) / 4.0;
AVFloatDSPContext *fdsp = venc->fdsp; AVFloatDSPContext *fdsp = venc->fdsp;
if (!venc->have_saved && !samples) for (channel = 0; channel < venc->channels; channel++) {
return 0; float *offset = venc->samples + channel * window_len * 2;
if (venc->have_saved) { fdsp->vector_fmul(offset, offset, win, samples);
for (channel = 0; channel < venc->channels; channel++) fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
memcpy(venc->samples + channel * window_len * 2,
venc->saved + channel * window_len, sizeof(float) * window_len);
} else {
for (channel = 0; channel < venc->channels; channel++)
memset(venc->samples + channel * window_len * 2, 0,
sizeof(float) * window_len);
}
if (samples) { offset += window_len;
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->samples + channel * window_len * 2 + window_len;
fdsp->vector_fmul_reverse(offset, audio + channel * window_len, win, samples); fdsp->vector_fmul_reverse(offset, offset, win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples); fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
}
} else {
for (channel = 0; channel < venc->channels; channel++)
memset(venc->samples + channel * window_len * 2 + window_len,
0, sizeof(float) * window_len);
}
for (channel = 0; channel < venc->channels; channel++)
venc->mdct[0].mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len, venc->mdct[0].mdct_calc(&venc->mdct[0], venc->coeffs + channel * window_len,
venc->samples + channel * window_len * 2); venc->samples + channel * window_len * 2);
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->saved + channel * window_len;
fdsp->vector_fmul(offset, audio + channel * window_len, win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
}
venc->have_saved = 1;
} else {
venc->have_saved = 0;
} }
return 1; return 1;
} }
@@ -1071,24 +1043,40 @@ static AVFrame *spawn_empty_frame(AVCodecContext *avctx, int channels)
return f; return f;
} }
/* Concatenate audio frames into an appropriately sized array of samples */ /* Set up audio samples for psy analysis and window/mdct */
static void move_audio(vorbis_enc_context *venc, float *audio, int *samples, int sf_size) static void move_audio(vorbis_enc_context *venc, int *samples, int sf_size)
{ {
AVFrame *cur = NULL; AVFrame *cur = NULL;
int frame_size = 1 << (venc->log2_blocksize[1] - 1); int frame_size = 1 << (venc->log2_blocksize[1] - 1);
int subframes = frame_size / sf_size; int subframes = frame_size / sf_size;
int sf, ch;
for (int sf = 0; sf < subframes; sf++) { /* Copy samples from last frame into current frame */
if (venc->have_saved)
for (ch = 0; ch < venc->channels; ch++)
memcpy(venc->samples + 2 * ch * frame_size,
venc->saved + ch * frame_size, sizeof(float) * frame_size);
else
for (ch = 0; ch < venc->channels; ch++)
memset(venc->samples + 2 * ch * frame_size, 0, sizeof(float) * frame_size);
for (sf = 0; sf < subframes; sf++) {
cur = ff_bufqueue_get(&venc->bufqueue); cur = ff_bufqueue_get(&venc->bufqueue);
*samples += cur->nb_samples; *samples += cur->nb_samples;
for (int ch = 0; ch < venc->channels; ch++) { for (ch = 0; ch < venc->channels; ch++) {
float *offset = venc->samples + 2 * ch * frame_size + frame_size;
float *save = venc->saved + ch * frame_size;
const float *input = (float *) cur->extended_data[ch]; const float *input = (float *) cur->extended_data[ch];
const size_t len = cur->nb_samples * sizeof(float); const size_t len = cur->nb_samples * sizeof(float);
memcpy(audio + ch*frame_size + sf*sf_size, input, len);
memcpy(offset + sf*sf_size, input, len);
memcpy(save + sf*sf_size, input, len); // Move samples for next frame
} }
av_frame_free(&cur); av_frame_free(&cur);
} }
venc->have_saved = 1;
memcpy(venc->scratch, venc->samples, 2 * venc->channels * frame_size);
} }
static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
@@ -1129,9 +1117,9 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
} }
} }
move_audio(venc, venc->scratch, &samples, avctx->frame_size); move_audio(venc, &samples, avctx->frame_size);
if (!apply_window_and_mdct(venc, venc->scratch, samples)) if (!apply_window_and_mdct(venc, samples))
return 0; return 0;
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0) if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)