Merge remote-tracking branch 'qatar/master'
* qatar/master: aac_latm: reconfigure decoder on audio specific config changes latmdec: fix audio specific config parsing Add avcodec_decode_audio4(). avcodec: change number of plane pointers from 4 to 8 at next major bump. Update developers documentation with coding conventions. svq1dec: avoid undefined get_bits(0) call ARM: h264dsp_neon cosmetics ARM: make some NEON macros reusable Do not memcpy raw video frames when using null muxer fate: update asf seektest vp8: flush buffers on size changes. doc: improve general documentation for MacOSX asf: use packet dts as approximation of pts asf: do not call av_read_frame rtsp: Initialize the media_type_mask in the rtp guessing demuxer Cleaned up alacenc.c Conflicts: doc/APIchanges doc/developer.texi libavcodec/8svx.c libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/nellymoserdec.c libavcodec/tta.c libavcodec/utils.c libavcodec/version.h libavcodec/wmadec.c libavformat/asfdec.c tests/ref/seek/lavf_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
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commit
e4de71677f
87 changed files with 2226 additions and 1307 deletions
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@ -507,20 +507,23 @@ static av_cold int sipr_decoder_init(AVCodecContext * avctx)
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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avcodec_get_frame_defaults(&ctx->frame);
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avctx->coded_frame = &ctx->frame;
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return 0;
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}
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static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
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int *data_size, AVPacket *avpkt)
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static int sipr_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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SiprContext *ctx = avctx->priv_data;
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const uint8_t *buf=avpkt->data;
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SiprParameters parm;
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const SiprModeParam *mode_par = &modes[ctx->mode];
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GetBitContext gb;
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float *data = datap;
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float *samples;
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int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
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int i, out_size;
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int i, ret;
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ctx->avctx = avctx;
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if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
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@ -530,27 +533,27 @@ static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
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return -1;
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}
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out_size = mode_par->frames_per_packet * subframe_size *
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mode_par->subframe_count *
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av_get_bytes_per_sample(avctx->sample_fmt);
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if (*data_size < out_size) {
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av_log(avctx, AV_LOG_ERROR,
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"Error processing packet: output buffer (%d) too small\n",
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*data_size);
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return -1;
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/* get output buffer */
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ctx->frame.nb_samples = mode_par->frames_per_packet * subframe_size *
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mode_par->subframe_count;
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if ((ret = avctx->get_buffer(avctx, &ctx->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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samples = (float *)ctx->frame.data[0];
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init_get_bits(&gb, buf, mode_par->bits_per_frame);
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for (i = 0; i < mode_par->frames_per_packet; i++) {
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decode_parameters(&parm, &gb, mode_par);
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ctx->decode_frame(ctx, &parm, data);
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ctx->decode_frame(ctx, &parm, samples);
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data += subframe_size * mode_par->subframe_count;
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samples += subframe_size * mode_par->subframe_count;
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}
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*data_size = out_size;
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*got_frame_ptr = 1;
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*(AVFrame *)data = ctx->frame;
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return mode_par->bits_per_frame >> 3;
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}
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@ -562,5 +565,6 @@ AVCodec ff_sipr_decoder = {
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.priv_data_size = sizeof(SiprContext),
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.init = sipr_decoder_init,
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.decode = sipr_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
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};
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