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aacdec: use float planar sample format for output

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pull/8/head
Justin Ruggles 12 years ago
parent 8e134e5104
commit 3d3cf6745e
3 changed files with 66 additions and 48 deletions
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  1. 7
      libavcodec/aac.h
  2. 101
      libavcodec/aacdec.c
  3. 6
      libavcodec/aacsbr.c

7
libavcodec/aac.h
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@ -236,9 +236,10 @@ typedef struct SingleChannelElement {
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output
DECLARE_ALIGNED(32, float, ret_buf)[2048]; ///< PCM output buffer
DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
float *ret; ///< PCM output
} SingleChannelElement;
/**
@ -297,10 +298,10 @@ typedef struct AACContext {
/** @} */
/**
* @name Members used for output interleaving
* @name Members used for output
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];

101
libavcodec/aacdec.c
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@ -149,10 +149,10 @@ static av_cold int che_configure(AACContext *ac,
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
if (type == TYPE_CPE ||
(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
}
}
} else {
@ -163,6 +163,38 @@ static av_cold int che_configure(AACContext *ac,
return 0;
}
static int frame_configure_elements(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
int type, id, ch, ret;
/* set channel pointers to internal buffers by default */
for (type = 0; type < 4; type++) {
for (id = 0; id < MAX_ELEM_ID; id++) {
ChannelElement *che = ac->che[type][id];
if (che) {
che->ch[0].ret = che->ch[0].ret_buf;
che->ch[1].ret = che->ch[1].ret_buf;
}
}
}
/* get output buffer */
ac->frame.nb_samples = 2048;
if ((ret = avctx->get_buffer(avctx, &ac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
if (ac->output_element[ch])
ac->output_element[ch]->ret = (float *)ac->frame.extended_data[ch];
}
return 0;
}
struct elem_to_channel {
uint64_t av_position;
uint8_t syn_ele;
@ -378,8 +410,8 @@ static void pop_output_configuration(AACContext *ac) {
* @return Returns error status. 0 - OK, !0 - error
*/
static int output_configure(AACContext *ac,
uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
enum OCStatus oc_type)
uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
enum OCStatus oc_type, int get_new_frame)
{
AVCodecContext *avctx = ac->avctx;
int i, channels = 0, ret;
@ -417,6 +449,11 @@ static int output_configure(AACContext *ac,
avctx->channels = ac->oc[1].channels = channels;
ac->oc[1].status = oc_type;
if (get_new_frame) {
if ((ret = frame_configure_elements(ac->avctx)) < 0)
return ret;
}
return 0;
}
@ -457,7 +494,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
2) < 0)
return NULL;
if (output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME) < 0)
OC_TRIAL_FRAME, 1) < 0)
return NULL;
ac->oc[1].m4ac.chan_config = 2;
@ -473,7 +510,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
1) < 0)
return NULL;
if (output_configure(ac, layout_map, layout_map_tags,
OC_TRIAL_FRAME) < 0)
OC_TRIAL_FRAME, 1) < 0)
return NULL;
ac->oc[1].m4ac.chan_config = 1;
@ -660,7 +697,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
} else if (m4ac->sbr == 1 && m4ac->ps == -1)
m4ac->ps = 1;
if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR)))
if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
return ret;
if (extension_flag) {
@ -802,11 +839,12 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
float output_scale_factor;
ac->avctx = avctx;
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
avctx->extradata,
@ -836,20 +874,12 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
&layout_map_tags, ac->oc[1].m4ac.chan_config);
if (!ret)
output_configure(ac, layout_map, layout_map_tags,
OC_GLOBAL_HDR);
OC_GLOBAL_HDR, 0);
else if (avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
}
if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
output_scale_factor = 1.0 / 32768.0;
} else {
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
output_scale_factor = 1.0;
}
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
AAC_INIT_VLC_STATIC( 2, 550);
@ -877,9 +907,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@ -1940,7 +1970,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
ac->oc[1].m4ac.sbr = 1;
ac->oc[1].m4ac.ps = 1;
output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
ac->oc[1].status);
ac->oc[1].status, 1);
} else {
ac->oc[1].m4ac.sbr = 1;
}
@ -2330,7 +2360,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
&layout_map_tags, hdr_info.chan_config))
return -7;
if (output_configure(ac, layout_map, layout_map_tags,
FFMAX(ac->oc[1].status, OC_TRIAL_FRAME)))
FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
return -7;
} else {
ac->oc[1].m4ac.chan_config = 0;
@ -2372,6 +2402,11 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
}
}
if (frame_configure_elements(avctx) < 0) {
err = -1;
goto fail;
}
ac->tags_mapped = 0;
// parse
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
@ -2426,7 +2461,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
pop_output_configuration(ac);
} else {
err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE);
err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
pce_found = 1;
}
break;
@ -2469,23 +2504,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
samples <<= multiplier;
if (samples) {
/* get output buffer */
ac->frame.nb_samples = samples;
if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
err = -1;
goto fail;
}
if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
else
ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
(const float **)ac->output_data,
samples, avctx->channels);
*(AVFrame *)data = ac->frame;
}
*got_frame_ptr = !!samples;
@ -2844,7 +2863,7 @@ AVCodec ff_aac_decoder = {
.decode = aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
@ -2865,7 +2884,7 @@ AVCodec ff_aac_latm_decoder = {
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,

6
libavcodec/aacsbr.c
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@ -140,7 +140,6 @@ static void sbr_turnoff(SpectralBandReplication *sbr) {
av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
{
float mdct_scale;
sbr->kx[0] = sbr->kx[1];
sbr_turnoff(sbr);
sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
@ -148,9 +147,8 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
/* SBR requires samples to be scaled to +/-32768.0 to work correctly.
* mdct scale factors are adjusted to scale up from +/-1.0 at analysis
* and scale back down at synthesis. */
mdct_scale = ac->avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? 32768.0f : 1.0f;
ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * mdct_scale));
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
ff_ps_ctx_init(&sbr->ps);
ff_sbrdsp_init(&sbr->dsp);
}

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