On Wed, Jan 18, 2012 at 12:39 AM, Nathan Caldwell <[email protected]> wrote:
> ---
> libavcodec/aacenc.c | 12 ++++++------
> libavcodec/aacenc.h | 2 +-
> libavcodec/aacpsy.c | 10 ++++++----
> libavcodec/psymodel.c | 7 +++----
> libavcodec/psymodel.h | 7 +++----
> 5 files changed, 19 insertions(+), 19 deletions(-)
>
> diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
> index 33396b2..3bd04bf 100644
> --- a/libavcodec/aacenc.c
> +++ b/libavcodec/aacenc.c
> @@ -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,
> };
> diff --git a/libavcodec/aacenc.h b/libavcodec/aacenc.h
> index acd185a..1f5e650 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;
> - 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
> diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
> index 577d8fb..adaab7a 100644
> --- a/libavcodec/aacpsy.c
> +++ b/libavcodec/aacpsy.c
> @@ -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 */
> @@ -808,6 +807,9 @@ 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]);
> }
> + /* NOTE: The LAME psymodel expects it's input in the range
> -32768 to 32768. Tuning this for normalized floats would be difficult. */
> + sum1 *= 32768.0f;
> + sum2 *= 32768.0f;
> hpfsmpl[i] = sum1 + sum2;
Why not hpfsmp[i] = 32768.0f * (sum1 + sum2); rather than scaling the
two individual sums?
> }
>
> diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
> index 740d859..49df118 100644
> --- a/libavcodec/psymodel.c
> +++ b/libavcodec/psymodel.c
> @@ -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 {
> diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
> index 3e866e4..03d078e 100644
> --- a/libavcodec/psymodel.h
> +++ b/libavcodec/psymodel.h
> @@ -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.
> --
> 1.7.5.4
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