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aacenc: Request normalized float samples instead of converting s16 samples to float.

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Signed-off-by: Alex Converse <alex.converse@gmail.com>
This commit is contained in:
Nathan Caldwell 2011-12-17 18:45:55 -07:00 committed by Alex Converse
parent 6381f913d1
commit 025ccf1f8b
5 changed files with 18 additions and 20 deletions
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12
libavcodec/aacenc.c
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@ -167,7 +167,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce, short *audio)
SingleChannelElement *sce, float *audio)
{
int i, k;
const int chans = avctx->channels;
@ -434,7 +434,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
int16_t *samples = s->samples, *samples2, *la;
float *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch;
int chan_el_counter[4];
@ -452,7 +452,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
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, (uint16_t*)data + start_ch,
ff_psy_preprocess(s->psypp, (float*)data + start_ch,
samples2 + start_ch, start_ch, chans);
start_ch += chans;
}
@ -621,9 +621,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows(7);
if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 1.0))
if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
return ret;
if (ret = ff_mdct_init(&s->mdct128, 8, 0, 1.0))
if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
return ret;
return 0;
@ -722,7 +722,7 @@ AVCodec ff_aac_encoder = {
.encode = aac_encode_frame,
.close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.priv_class = &aacenc_class,
};

2
libavcodec/aacenc.h
View File

@ -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;
int16_t *samples; ///< saved preprocessed input
float *samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
const uint8_t *chan_map; ///< channel configuration map

10
libavcodec/aacpsy.c
View File

@ -776,9 +776,8 @@ static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int u
ctx->next_window_seq = blocktype;
}
static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
const float *la, int channel, int prev_type)
{
AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
AacPsyChannel *pch = &pctx->ch[channel];
@ -796,7 +795,7 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
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 int16_t *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) * chans;
int j, att_sum = 0;
/* LAME comment: apply high pass filter of fs/4 */
@ -808,7 +807,8 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
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]);
}
hpfsmpl[i] = sum1 + sum2;
/* 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;
}
/* Calculate the energies of each sub-shortblock */

7
libavcodec/psymodel.c
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@ -112,14 +112,13 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
return ctx;
}
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels)
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, const float *audio,
float *dest, int tag, int channels)
{
int ch, i;
if (ctx->fstate) {
for (ch = 0; ch < channels; ch++)
ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
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 {

7
libavcodec/psymodel.h
View File

@ -109,7 +109,7 @@ typedef struct FFPsyModel {
*
* @return suggested window information in a structure
*/
FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
FFPsyWindowInfo (*window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type);
/**
* Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels.
@ -179,9 +179,8 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
* @param tag channel number
* @param channels number of channel to preprocess (some additional work may be done on stereo pair)
*/
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels);
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, const float *audio,
float *dest, int tag, int channels);
/**
* Cleanup audio preprocessing module.