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https://github.com/FFmpeg/FFmpeg.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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@ -25,6 +25,10 @@
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#define AES3_HEADER_LEN 4
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typedef struct S302MDecodeContext {
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AVFrame frame;
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} S302MDecodeContext;
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static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
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int buf_size)
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{
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@ -83,10 +87,12 @@ static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
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}
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static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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int *data_size, AVPacket *avpkt)
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int *got_frame_ptr, AVPacket *avpkt)
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{
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S302MDecodeContext *s = avctx->priv_data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int block_size, ret;
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int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
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if (frame_size < 0)
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@ -95,11 +101,18 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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buf_size -= AES3_HEADER_LEN;
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buf += AES3_HEADER_LEN;
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if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
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return -1;
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/* get output buffer */
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block_size = (avctx->bits_per_coded_sample + 4) / 4;
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s->frame.nb_samples = 2 * (buf_size / block_size) / avctx->channels;
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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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buf_size = (s->frame.nb_samples * avctx->channels / 2) * block_size;
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if (avctx->bits_per_coded_sample == 24) {
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uint32_t *o = data;
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uint32_t *o = (uint32_t *)s->frame.data[0];
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for (; buf_size > 6; buf_size -= 7) {
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*o++ = (av_reverse[buf[2]] << 24) |
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(av_reverse[buf[1]] << 16) |
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@ -110,9 +123,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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(av_reverse[buf[3] & 0x0f] << 4);
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buf += 7;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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} else if (avctx->bits_per_coded_sample == 20) {
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uint32_t *o = data;
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uint32_t *o = (uint32_t *)s->frame.data[0];
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for (; buf_size > 5; buf_size -= 6) {
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*o++ = (av_reverse[buf[2] & 0xf0] << 28) |
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(av_reverse[buf[1]] << 20) |
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@ -122,9 +134,8 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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(av_reverse[buf[3]] << 12);
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buf += 6;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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} else {
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uint16_t *o = data;
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uint16_t *o = (uint16_t *)s->frame.data[0];
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for (; buf_size > 4; buf_size -= 5) {
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*o++ = (av_reverse[buf[1]] << 8) |
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av_reverse[buf[0]];
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@ -133,10 +144,22 @@ static int s302m_decode_frame(AVCodecContext *avctx, void *data,
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(av_reverse[buf[2]] >> 4);
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buf += 5;
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}
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*data_size = (uint8_t*) o - (uint8_t*) data;
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}
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return buf - avpkt->data;
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return avpkt->size;
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}
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static int s302m_decode_init(AVCodecContext *avctx)
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{
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S302MDecodeContext *s = avctx->priv_data;
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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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}
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@ -144,6 +167,9 @@ AVCodec ff_s302m_decoder = {
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.name = "s302m",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_S302M,
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.priv_data_size = sizeof(S302MDecodeContext),
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.init = s302m_decode_init,
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.decode = s302m_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
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};
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