mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
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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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@@ -129,6 +129,7 @@ typedef struct APEPredictor {
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/** Decoder context */
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typedef struct APEContext {
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AVCodecContext *avctx;
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AVFrame frame;
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DSPContext dsp;
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int channels;
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int samples; ///< samples left to decode in current frame
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@@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
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dsputil_init(&s->dsp, avctx);
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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filter_alloc_fail:
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ape_decode_close(avctx);
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@@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
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}
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}
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static int ape_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt)
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static int ape_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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APEContext *s = avctx->priv_data;
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int16_t *samples = data;
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int i;
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int blockstodecode, out_size;
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int16_t *samples;
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int i, ret;
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int blockstodecode;
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int bytes_used = 0;
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/* this should never be negative, but bad things will happen if it is, so
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@@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx,
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void *tmp_data;
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if (!buf_size) {
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*data_size = 0;
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*got_frame_ptr = 0;
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return 0;
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}
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if (buf_size < 8) {
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@@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx,
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}
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if (!s->data) {
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*data_size = 0;
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*got_frame_ptr = 0;
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return buf_size;
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}
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blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples);
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out_size = blockstodecode * avctx->channels *
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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, "Output buffer is too small.\n");
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return AVERROR(EINVAL);
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/* get output buffer */
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s->frame.nb_samples = blockstodecode;
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if ((ret = avctx->get_buffer(avctx, &s->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 = (int16_t *)s->frame.data[0];
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s->error=0;
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@@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx,
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s->samples -= blockstodecode;
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*data_size = out_size;
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return bytes_used;
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}
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@@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = {
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.init = ape_decode_init,
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.close = ape_decode_close,
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.decode = ape_decode_frame,
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.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
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.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
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.flush = ape_flush,
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.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
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};
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