if (che_pos[type][id]) {
if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
- ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+ 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;
if (type == TYPE_CPE ||
reset_predict_state(&ps[i]);
}
+static int sample_rate_idx (int rate)
+{
+ if (92017 <= rate) return 0;
+ else if (75132 <= rate) return 1;
+ else if (55426 <= rate) return 2;
+ else if (46009 <= rate) return 3;
+ else if (37566 <= rate) return 4;
+ else if (27713 <= rate) return 5;
+ else if (23004 <= rate) return 6;
+ else if (18783 <= rate) return 7;
+ else if (13856 <= rate) return 8;
+ else if (11502 <= rate) return 9;
+ else if (9391 <= rate) return 10;
+ else return 11;
+}
+
static void reset_predictor_group(PredictorState *ps, int group_num)
{
int i;
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
+ float output_scale_factor;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
avctx->extradata,
avctx->extradata_size) < 0)
return -1;
+ } else {
+ int sr, i;
+ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+
+ sr = sample_rate_idx(avctx->sample_rate);
+ ac->m4ac.sampling_index = sr;
+ ac->m4ac.channels = avctx->channels;
+ ac->m4ac.sbr = -1;
+ ac->m4ac.ps = -1;
+
+ for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+ if (ff_mpeg4audio_channels[i] == avctx->channels)
+ break;
+ if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+ i = 0;
+ }
+ ac->m4ac.chan_config = i;
+
+ if (ac->m4ac.chan_config) {
+ int ret = set_default_channel_config(avctx, new_che_pos, ac->m4ac.chan_config);
+ if (!ret)
+ output_configure(ac, ac->che_pos, new_che_pos, ac->m4ac.chan_config, OC_GLOBAL_HDR);
+ else if (avctx->error_recognition >= FF_ER_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ 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);
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
- ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
- ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
- ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
+ 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);
// 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);
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
clipped_offset = av_clip(offset[1], -100, 155);
- if (offset[2] != clipped_offset) {
+ if (offset[1] != clipped_offset) {
av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
"(%d -> %d).\nIf you heard an audible "
"artifact, there may be a bug in the decoder. ",
} else {
memset(in, 0, 448 * sizeof(float));
ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
- memcpy(in + 576, in + 576, 448 * sizeof(float));
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
- memcpy(in + 1024, in + 1024, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(float));
}
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
}
- memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 * sizeof(int16_t));
- ac->fmt_conv.float_to_int16(&(sce->ltp_state[1024]), sce->ret, 1024);
- ac->fmt_conv.float_to_int16(&(sce->ltp_state[2048]), saved_ltp, 1024);
+ memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
}
/**
if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
return -7;
} else if (ac->output_configured != OC_LOCKED) {
+ ac->m4ac.chan_config = 0;
ac->output_configured = OC_NONE;
}
if (ac->output_configured != OC_LOCKED) {
ac->m4ac.sbr = -1;
ac->m4ac.ps = -1;
+ ac->m4ac.sample_rate = hdr_info.sample_rate;
+ ac->m4ac.sampling_index = hdr_info.sampling_index;
+ ac->m4ac.object_type = hdr_info.object_type;
}
- ac->m4ac.sample_rate = hdr_info.sample_rate;
- ac->m4ac.sampling_index = hdr_info.sampling_index;
- ac->m4ac.object_type = hdr_info.object_type;
if (!ac->avctx->sample_rate)
ac->avctx->sample_rate = hdr_info.sample_rate;
if (hdr_info.num_aac_frames == 1) {
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
int err, elem_id, data_size_tmp;
- int samples = 0, multiplier;
+ int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, gb) < 0) {
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
+ audio_found = 1;
break;
case TYPE_CCE:
case TYPE_LFE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_DSE:
avctx->frame_size = samples;
}
- data_size_tmp = samples * avctx->channels * sizeof(int16_t);
+ data_size_tmp = samples * avctx->channels *
+ av_get_bytes_per_sample(avctx->sample_fmt);
if (*data_size < data_size_tmp) {
av_log(avctx, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
}
*data_size = data_size_tmp;
- if (samples)
- ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+ if (samples) {
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+ ac->fmt_conv.float_interleave(data, (const float **)ac->output_data,
+ samples, avctx->channels);
+ else
+ ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data,
+ samples, avctx->channels);
+ }
- if (ac->output_configured)
+ if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
return 0;
*out_size = 0;
return avpkt->size;
} else {
+ aac_decode_close(avctx);
if ((err = aac_decode_init(avctx)) < 0)
return err;
latmctx->initialized = 1;
AVCodec ff_aac_decoder = {
- "aac",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_AAC,
- sizeof(AACContext),
- aac_decode_init,
- NULL,
- aac_decode_close,
- aac_decode_frame,
+ .name = "aac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACContext),
+ .init = aac_decode_init,
+ .close = aac_decode_close,
+ .decode = aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF,
.channel_layouts = aac_channel_layout,
};
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF,
.channel_layouts = aac_channel_layout,
};
projects / ffmpeg / blobdiff