[FFmpeg-cvslog] aacenc: Deinterleave input samples before processing.
Nathan Caldwell
git at videolan.org
Tue Jan 24 03:04:45 CET 2012
ffmpeg | branch: master | Nathan Caldwell <saintdev at gmail.com> | Wed Dec 14 19:43:56 2011 -0700| [9b8e2a870957293898998209c6e9bed290cc9bef] | committer: Alex Converse
aacenc: Deinterleave input samples before processing.
Signed-off-by: Alex Converse <alex.converse at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=9b8e2a870957293898998209c6e9bed290cc9bef
---
libavcodec/aacenc.c | 100 ++++++++++++++++++++++++++++++-------------------
libavcodec/aacenc.h | 6 ++-
libavcodec/aacpsy.c | 11 ++---
libavcodec/psymodel.c | 16 +++-----
libavcodec/psymodel.h | 9 +---
5 files changed, 80 insertions(+), 62 deletions(-)
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index a25edeb..742fee9 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -144,6 +144,18 @@ static const uint8_t aac_chan_configs[6][5] = {
};
/**
+ * Table to remap channels from Libav's default order to AAC order.
+ */
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+ { 0 },
+ { 0, 1 },
+ { 2, 0, 1 },
+ { 2, 0, 1, 3 },
+ { 2, 0, 1, 3, 4 },
+ { 2, 0, 1, 4, 5, 3 },
+};
+
+/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
@@ -172,34 +184,29 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
float *audio)
{
int i, k;
- const int chans = s->channels;
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float *output = sce->ret;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
- memcpy(output, sce->saved, sizeof(float)*1024);
+ memcpy(output, sce->saved, sizeof(output[0])*1024);
if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
memset(output, 0, sizeof(output[0]) * 448);
for (i = 448; i < 576; i++)
output[i] = sce->saved[i] * pwindow[i - 448];
- for (i = 576; i < 704; i++)
- output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
for (i = 0; i < 1024; i++) {
- output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
- sce->saved[i] = audio[i * chans] * lwindow[i];
+ output[i+1024] = audio[i] * lwindow[1024 - i - 1];
+ sce->saved[i] = audio[i] * lwindow[i];
}
} else {
- for (i = 0; i < 448; i++)
- output[i+1024] = audio[i * chans];
+ memcpy(output + 1024, audio, sizeof(output[0]) * 448);
for (; i < 576; i++)
- output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
+ output[i+1024] = audio[i] * swindow[576 - i - 1];
memset(output+1024+576, 0, sizeof(output[0]) * 448);
- for (i = 0; i < 1024; i++)
- sce->saved[i] = audio[i * chans];
+ memcpy(sce->saved, audio, sizeof(sce->saved[0]) * 1024);
}
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
} else {
@@ -207,13 +214,12 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
for (i = 448 + k; i < 448 + k + 256; i++)
output[i - 448 - k] = (i < 1024)
? sce->saved[i]
- : audio[(i-1024)*chans];
+ : audio[i-1024];
s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
}
- for (i = 0; i < 1024; i++)
- sce->saved[i] = audio[i * chans];
+ memcpy(sce->saved, audio, sizeof(sce->saved[0]) * 1024);
}
}
@@ -432,11 +438,37 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
put_bits(&s->pb, 12 - padbits, 0);
}
+/*
+ * Deinterleave input samples.
+ * Channels are reordered from Libav's default order to AAC order.
+ */
+static void deinterleave_input_samples(AACEncContext *s,
+ const float *samples)
+{
+ int ch, i;
+ const int sinc = s->channels;
+ const uint8_t *channel_map = aac_chan_maps[sinc - 1];
+
+ /* deinterleave and remap input samples */
+ for (ch = 0; ch < sinc; ch++) {
+ const float *sptr = samples + channel_map[ch];
+
+ /* copy last 1024 samples of previous frame to the start of the current frame */
+ memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0]));
+
+ /* deinterleave */
+ for (i = 1024; i < 1024 * 2; i++) {
+ s->planar_samples[ch][i] = *sptr;
+ sptr += sinc;
+ }
+ }
+}
+
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
- float *samples = s->samples, *samples2, *la;
+ float **samples = s->planar_samples, *samples2, *la;
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch;
int chan_el_counter[4];
@@ -444,27 +476,15 @@ static int aac_encode_frame(AVCodecContext *avctx,
if (s->last_frame)
return 0;
+
if (data) {
- if (!s->psypp) {
- memcpy(s->samples + 1024 * s->channels, data,
- 1024 * s->channels * sizeof(s->samples[0]));
- } else {
- start_ch = 0;
- samples2 = s->samples + 1024 * s->channels;
- for (i = 0; i < s->chan_map[0]; i++) {
- tag = s->chan_map[i+1];
- chans = tag == TYPE_CPE ? 2 : 1;
- ff_psy_preprocess(s->psypp, (float*)data + start_ch,
- samples2 + start_ch, start_ch, chans);
- start_ch += chans;
- }
- }
+ deinterleave_input_samples(s, data);
+ if (s->psypp)
+ ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
}
- if (!avctx->frame_number) {
- memcpy(s->samples, s->samples + 1024 * s->channels,
- 1024 * s->channels * sizeof(s->samples[0]));
+
+ if (!avctx->frame_number)
return 0;
- }
start_ch = 0;
for (i = 0; i < s->chan_map[0]; i++) {
@@ -475,8 +495,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
int cur_channel = start_ch + ch;
- samples2 = samples + cur_channel;
- la = samples2 + (448+64) * s->channels;
+ samples2 = &samples[cur_channel][0];
+ la = samples2 + (448+64);
if (!data)
la = NULL;
if (tag == TYPE_LFE) {
@@ -592,8 +612,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
if (!data)
s->last_frame = 1;
- memcpy(s->samples, s->samples + 1024 * s->channels,
- 1024 * s->channels * sizeof(s->samples[0]));
+
return put_bits_count(&s->pb)>>3;
}
@@ -606,7 +625,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
ff_psy_end(&s->psy);
if (s->psypp)
ff_psy_preprocess_end(s->psypp);
- av_freep(&s->samples);
+ av_freep(&s->buffer.samples);
av_freep(&s->cpe);
return 0;
}
@@ -633,10 +652,13 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
{
- FF_ALLOC_OR_GOTO (avctx, s->samples, 2 * 1024 * s->channels * sizeof(s->samples[0]), alloc_fail);
+ FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 2 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+ for(int ch = 0; ch < s->channels; ch++)
+ s->planar_samples[ch] = s->buffer.samples + 2 * 1024 * ch;
+
return 0;
alloc_fail:
return AVERROR(ENOMEM);
diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
index 211c70a..44ab13b 100644
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -58,7 +58,7 @@ typedef struct AACEncContext {
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
- float *samples; ///< saved preprocessed input
+ float *planar_samples[6]; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
int channels; ///< channel count
@@ -73,6 +73,10 @@ typedef struct AACEncContext {
float lambda;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
+
+ struct {
+ float *samples;
+ } buffer;
} AACEncContext;
extern float ff_aac_pow34sf_tab[428];
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 5e9e391..8ee393a 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -400,7 +400,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
int stay_short = 0;
for (i = 0; i < 8; i++) {
for (j = 0; j < 128; j++) {
- v = iir_filter(la[(i*128+j)*ctx->avctx->channels], pch->iir_state);
+ v = iir_filter(la[i*128+j], pch->iir_state);
sum += v*v;
}
s[i] = sum;
@@ -794,18 +794,17 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
- int chans = ctx->avctx->channels;
- const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN) * chans;
+ const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
int j, att_sum = 0;
/* LAME comment: apply high pass filter of fs/4 */
for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
float sum1, sum2;
- sum1 = firbuf[(i + ((PSY_LAME_FIR_LEN - 1) / 2)) * chans];
+ sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
sum2 = 0.0;
for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
- sum1 += psy_fir_coeffs[j] * (firbuf[(i + j) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j) * chans]);
- sum2 += psy_fir_coeffs[j + 1] * (firbuf[(i + j + 1) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j - 1) * chans]);
+ sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
+ sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
}
/* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. Tuning this for normalized floats would be difficult. */
hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index 49df118..316076a 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -112,19 +112,15 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
return ctx;
}
-void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, const float *audio,
- float *dest, int tag, int channels)
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
{
- int ch, i;
+ int ch;
+ int frame_size = ctx->avctx->frame_size;
+
if (ctx->fstate) {
for (ch = 0; ch < channels; ch++)
- ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
- audio + ch, ctx->avctx->channels,
- dest + ch, ctx->avctx->channels);
- } else {
- for (ch = 0; ch < channels; ch++)
- for (i = 0; i < ctx->avctx->frame_size; i++)
- dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
+ ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
+ &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
}
}
diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
index 03d078e..34b20d7 100644
--- a/libavcodec/psymodel.h
+++ b/libavcodec/psymodel.h
@@ -174,13 +174,10 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
* Preprocess several channel in audio frame in order to compress it better.
*
* @param ctx preprocessing context
- * @param audio samples to preprocess
- * @param dest place to put filtered samples
- * @param tag channel number
- * @param channels number of channel to preprocess (some additional work may be done on stereo pair)
+ * @param audio samples to be filtered (in place)
+ * @param channels number of channel to preprocess
*/
-void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, const float *audio,
- float *dest, int tag, int channels);
+void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels);
/**
* Cleanup audio preprocessing module.
More information about the ffmpeg-cvslog
mailing list