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https://github.com/xenia-project/FFmpeg.git
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Merge commit '3d3cf6745e2a5dc9c377244454c3186d75b177fa'
* commit '3d3cf6745e2a5dc9c377244454c3186d75b177fa': aacdec: use float planar sample format for output Conflicts: libavcodec/aacdec.c libavcodec/aacsbr.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
commit
59b68ee887
@ -236,9 +236,10 @@ typedef struct SingleChannelElement {
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uint8_t zeroes[128]; ///< band is not coded (used by encoder)
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DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT
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DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap
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DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output
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DECLARE_ALIGNED(32, float, ret_buf)[2048]; ///< PCM output buffer
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DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
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PredictorState predictor_state[MAX_PREDICTORS];
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float *ret; ///< PCM output
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} SingleChannelElement;
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/**
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@ -297,10 +298,10 @@ typedef struct AACContext {
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/** @} */
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/**
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* @name Members used for output interleaving
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* @name Members used for output
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* @{
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*/
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float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
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/** @} */
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@ -153,10 +153,10 @@ static av_cold int che_configure(AACContext *ac,
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av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
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return AVERROR_INVALIDDATA;
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}
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
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ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
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if (type == TYPE_CPE ||
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(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
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ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
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}
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}
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} else {
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@ -167,6 +167,38 @@ static av_cold int che_configure(AACContext *ac,
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return 0;
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}
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static int frame_configure_elements(AVCodecContext *avctx)
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{
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AACContext *ac = avctx->priv_data;
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int type, id, ch, ret;
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/* set channel pointers to internal buffers by default */
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for (type = 0; type < 4; type++) {
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for (id = 0; id < MAX_ELEM_ID; id++) {
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ChannelElement *che = ac->che[type][id];
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if (che) {
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che->ch[0].ret = che->ch[0].ret_buf;
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che->ch[1].ret = che->ch[1].ret_buf;
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}
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}
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}
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/* get output buffer */
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ac->frame.nb_samples = 2048;
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if ((ret = avctx->get_buffer(avctx, &ac->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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/* map output channel pointers to AVFrame data */
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for (ch = 0; ch < avctx->channels; ch++) {
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if (ac->output_element[ch])
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ac->output_element[ch]->ret = (float *)ac->frame.extended_data[ch];
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}
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return 0;
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}
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struct elem_to_channel {
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uint64_t av_position;
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uint8_t syn_ele;
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@ -382,8 +414,8 @@ static void pop_output_configuration(AACContext *ac) {
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int output_configure(AACContext *ac,
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uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
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enum OCStatus oc_type)
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uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
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enum OCStatus oc_type, int get_new_frame)
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{
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AVCodecContext *avctx = ac->avctx;
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int i, channels = 0, ret;
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@ -422,6 +454,11 @@ static int output_configure(AACContext *ac,
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avctx->channels = ac->oc[1].channels = channels;
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ac->oc[1].status = oc_type;
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if (get_new_frame) {
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if ((ret = frame_configure_elements(ac->avctx)) < 0)
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return ret;
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}
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return 0;
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}
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@ -481,7 +518,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
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2) < 0)
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return NULL;
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if (output_configure(ac, layout_map, layout_map_tags,
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OC_TRIAL_FRAME) < 0)
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OC_TRIAL_FRAME, 1) < 0)
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return NULL;
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ac->oc[1].m4ac.chan_config = 2;
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@ -499,7 +536,7 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
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1) < 0)
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return NULL;
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if (output_configure(ac, layout_map, layout_map_tags,
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OC_TRIAL_FRAME) < 0)
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OC_TRIAL_FRAME, 1) < 0)
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return NULL;
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ac->oc[1].m4ac.chan_config = 1;
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@ -692,7 +729,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
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} else if (m4ac->sbr == 1 && m4ac->ps == -1)
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m4ac->ps = 1;
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if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR)))
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if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
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return ret;
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if (extension_flag) {
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@ -834,18 +871,11 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
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static av_cold int aac_decode_init(AVCodecContext *avctx)
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{
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AACContext *ac = avctx->priv_data;
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float output_scale_factor;
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ac->avctx = avctx;
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ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
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if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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output_scale_factor = 1.0 / 32768.0;
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} else {
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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output_scale_factor = 1.0;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
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if (avctx->extradata_size > 0) {
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if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
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@ -876,7 +906,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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&layout_map_tags, ac->oc[1].m4ac.chan_config);
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if (!ret)
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output_configure(ac, layout_map, layout_map_tags,
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OC_GLOBAL_HDR);
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OC_GLOBAL_HDR, 0);
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else if (avctx->err_recognition & AV_EF_EXPLODE)
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return AVERROR_INVALIDDATA;
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}
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@ -909,9 +939,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
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ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
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352);
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ff_mdct_init(&ac->mdct, 11, 1, output_scale_factor/1024.0);
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ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
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ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
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ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
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ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
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ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
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// window initialization
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ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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@ -2001,7 +2031,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
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ac->oc[1].m4ac.sbr = 1;
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ac->oc[1].m4ac.ps = 1;
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output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
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ac->oc[1].status);
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ac->oc[1].status, 1);
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} else {
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ac->oc[1].m4ac.sbr = 1;
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}
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@ -2395,7 +2425,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
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&layout_map_tags, hdr_info.chan_config))
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return -7;
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if (output_configure(ac, layout_map, layout_map_tags,
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FFMAX(ac->oc[1].status, OC_TRIAL_FRAME)))
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FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
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return -7;
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} else {
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ac->oc[1].m4ac.chan_config = 0;
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@ -2404,6 +2434,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
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* WITHOUT specifying PCE.
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* thus, set dual mono as default.
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*/
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#if 0
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if (ac->enable_jp_dmono && ac->oc[0].status == OC_NONE) {
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layout_map_tags = 2;
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layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
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@ -2414,6 +2445,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
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OC_TRIAL_FRAME))
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return -7;
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}
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#endif
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}
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ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
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ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
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@ -2454,6 +2486,11 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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}
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}
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if (frame_configure_elements(avctx) < 0) {
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err = -1;
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goto fail;
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}
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ac->tags_mapped = 0;
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// parse
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while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
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@ -2509,7 +2546,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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"Not evaluating a further program_config_element as this construct is dubious at best.\n");
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pop_output_configuration(ac);
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} else {
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err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE);
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err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
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if (!err)
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ac->oc[1].m4ac.chan_config = 0;
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pce_found = 1;
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@ -2552,7 +2589,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
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samples <<= multiplier;
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#if 0
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/* for dual-mono audio (SCE + SCE) */
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is_dmono = ac->enable_jp_dmono && sce_count == 2 &&
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ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
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@ -2566,36 +2603,20 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
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ac->output_data[0] = ac->output_data[1];
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}
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}
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#endif
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if (samples) {
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/* get output buffer */
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ac->frame.nb_samples = samples;
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if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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err = -1;
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goto fail;
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}
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if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
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ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
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(const float **)ac->output_data,
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samples, avctx->channels);
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else
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ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
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(const float **)ac->output_data,
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samples, avctx->channels);
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*(AVFrame *)data = ac->frame;
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}
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*got_frame_ptr = !!samples;
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#if 0
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if (is_dmono) {
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if (ac->dmono_mode == 0)
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ac->output_data[1] = tmp;
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else if (ac->dmono_mode == 1)
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ac->output_data[0] = tmp;
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}
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#endif
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if (ac->oc[1].status && audio_found) {
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avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
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avctx->frame_size = samples;
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@ -2970,7 +2991,7 @@ AVCodec ff_aac_decoder = {
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.decode = aac_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
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.sample_fmts = (const enum AVSampleFormat[]) {
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
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},
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.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
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.channel_layouts = aac_channel_layout,
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@ -2992,7 +3013,7 @@ AVCodec ff_aac_latm_decoder = {
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.decode = latm_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
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.sample_fmts = (const enum AVSampleFormat[]) {
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
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},
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.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
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.channel_layouts = aac_channel_layout,
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@ -191,7 +191,7 @@ WINDOW_FUNC(only_long)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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float *out = sce->ret;
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float *out = sce->ret_buf;
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fdsp->vector_fmul (out, audio, lwindow, 1024);
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dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
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@ -201,7 +201,7 @@ WINDOW_FUNC(long_start)
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{
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const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret;
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float *out = sce->ret_buf;
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fdsp->vector_fmul(out, audio, lwindow, 1024);
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memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
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@ -213,7 +213,7 @@ WINDOW_FUNC(long_stop)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret;
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float *out = sce->ret_buf;
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memset(out, 0, sizeof(out[0]) * 448);
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fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
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@ -226,7 +226,7 @@ WINDOW_FUNC(eight_short)
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *in = audio + 448;
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float *out = sce->ret;
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float *out = sce->ret_buf;
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int w;
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for (w = 0; w < 8; w++) {
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@ -251,7 +251,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
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float *audio)
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{
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int i;
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float *output = sce->ret;
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float *output = sce->ret_buf;
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apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio);
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@ -142,7 +142,6 @@ static void sbr_turnoff(SpectralBandReplication *sbr) {
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av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
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{
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float mdct_scale;
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if(sbr->mdct.mdct_bits)
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return;
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sbr->kx[0] = sbr->kx[1];
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@ -152,9 +151,8 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
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/* SBR requires samples to be scaled to +/-32768.0 to work correctly.
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* mdct scale factors are adjusted to scale up from +/-1.0 at analysis
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* and scale back down at synthesis. */
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mdct_scale = ac->avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? 32768.0f : 1.0f;
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ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * mdct_scale));
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ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
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ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
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ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
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ff_ps_ctx_init(&sbr->ps);
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ff_sbrdsp_init(&sbr->dsp);
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}
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