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authorMichael Niedermayer <michaelni@gmx.at>2011-05-12 04:51:24 +0200
committerMichael Niedermayer <michaelni@gmx.at>2011-05-12 04:51:24 +0200
commit612122b187d711257eecd517e4049cef3bb0b7f0 (patch)
tree2e0ed86f6f73bbc993a0e7787f331e21d1c7c064 /libavcodec/s302m.c
parent4ea216e761e02d3f6973b316feaf3484be91a14f (diff)
parent5705b02079449c685a3dd337fcc3a8b440dca4a0 (diff)
downloadffmpeg-612122b187d711257eecd517e4049cef3bb0b7f0.tar.gz
Merge remote branch 'qatar/master'
* qatar/master: (32 commits) 10-bit H.264 x86 chroma v loopfilter asm Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected] Fix crash of interlaced MPEG2 decoding h264pred: fix one more aliasing violation. doc/APIchanges: fill in missing hashes and dates. flacenc: use proper initializers for AVOption default values. lavc: deprecate named constants for deprecated antialias_algo. aac: workaround for compilation on cygwin swscale: extend YUV422p support to 10bits depth tiff: add support for inverted FillOrder for uncompressed data Remove unused softfloat implementation. h264pred: fix aliasing violations. rotozoom: Eliminate French variable name. rotozoom: Check return value of fread(). rotozoom: Return an error value instead of calling exit(). rotozoom: Make init_demo() return int and check for errors on invocation. rotozoom: Drop silly UINT8 typedef. rotozoom: Drop some unnecessary parentheses. rotozoom: K&R coding style cosmetics rtsp: Only do keepalive using GET_PARAMETER if the server supports it ... Conflicts: Changelog cmdutils.c doc/APIchanges doc/general.texi ffmpeg.c ffplay.c libavcodec/h264pred_template.c libavcodec/resample.c libavutil/pixfmt.h libavutil/softfloat.c libavutil/softfloat.h tests/rotozoom.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/s302m.c')
-rw-r--r--libavcodec/s302m.c141
1 files changed, 141 insertions, 0 deletions
diff --git a/libavcodec/s302m.c b/libavcodec/s302m.c
new file mode 100644
index 0000000000..dd0ec2ee19
--- /dev/null
+++ b/libavcodec/s302m.c
@@ -0,0 +1,141 @@
+/*
+ * SMPTE 302M decoder
+ * Copyright (c) 2008 Laurent Aimar <fenrir@videolan.org>
+ * Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+
+#define AES3_HEADER_LEN 4
+
+static int s302m_parse_frame_header(AVCodecContext *avctx, const uint8_t *buf,
+ int buf_size)
+{
+ uint32_t h;
+ int frame_size, channels, id, bits;
+
+ if (buf_size <= AES3_HEADER_LEN) {
+ av_log(avctx, AV_LOG_ERROR, "frame is too short\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ /*
+ * AES3 header :
+ * size: 16
+ * number channels 2
+ * channel_id 8
+ * bits per samples 2
+ * alignments 4
+ */
+
+ h = AV_RB32(buf);
+ frame_size = (h >> 16) & 0xffff;
+ channels = ((h >> 14) & 0x0003) * 2 + 2;
+ id = (h >> 6) & 0x00ff;
+ bits = ((h >> 4) & 0x0003) * 4 + 16;
+
+ if (AES3_HEADER_LEN + frame_size != buf_size || bits > 24) {
+ av_log(avctx, AV_LOG_ERROR, "frame has invalid header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* Set output properties */
+ avctx->bits_per_coded_sample = bits;
+ if (bits > 16)
+ avctx->sample_fmt = SAMPLE_FMT_S32;
+ else
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+
+ avctx->channels = channels;
+ avctx->sample_rate = 48000;
+ avctx->bit_rate = 48000 * avctx->channels * (avctx->bits_per_coded_sample + 4) +
+ 32 * (48000 / (buf_size * 8 /
+ (avctx->channels *
+ (avctx->bits_per_coded_sample + 4))));
+
+ return frame_size;
+}
+
+static int s302m_decode_frame(AVCodecContext *avctx, void *data,
+ int *data_size, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+
+ int frame_size = s302m_parse_frame_header(avctx, buf, buf_size);
+ if (frame_size < 0)
+ return frame_size;
+
+ buf_size -= AES3_HEADER_LEN;
+ buf += AES3_HEADER_LEN;
+
+ if (*data_size < 4 * buf_size * 8 / (avctx->bits_per_coded_sample + 4))
+ return -1;
+
+ if (avctx->bits_per_coded_sample == 24) {
+ uint32_t *o = data;
+ for (; buf_size > 6; buf_size -= 7) {
+ *o++ = (av_reverse[buf[2]] << 24) |
+ (av_reverse[buf[1]] << 16) |
+ (av_reverse[buf[0]] << 8);
+ *o++ = (av_reverse[buf[6] & 0xf0] << 28) |
+ (av_reverse[buf[5]] << 20) |
+ (av_reverse[buf[4]] << 12) |
+ (av_reverse[buf[3] & 0x0f] << 8);
+ buf += 7;
+ }
+ *data_size = (uint8_t*) o - (uint8_t*) data;
+ } else if (avctx->bits_per_coded_sample == 20) {
+ uint32_t *o = data;
+ for (; buf_size > 5; buf_size -= 6) {
+ *o++ = (av_reverse[buf[2] & 0xf0] << 28) |
+ (av_reverse[buf[1]] << 20) |
+ (av_reverse[buf[0]] << 12);
+ *o++ = (av_reverse[buf[5] & 0xf0] << 28) |
+ (av_reverse[buf[4]] << 20) |
+ (av_reverse[buf[3]] << 12);
+ buf += 6;
+ }
+ *data_size = (uint8_t*) o - (uint8_t*) data;
+ } else {
+ uint16_t *o = data;
+ for (; buf_size > 4; buf_size -= 5) {
+ *o++ = (av_reverse[buf[1]] << 8) |
+ av_reverse[buf[0]];
+ *o++ = (av_reverse[buf[4] & 0xf0] << 12) |
+ (av_reverse[buf[3]] << 4) |
+ av_reverse[buf[2] & 0x0f];
+ buf += 5;
+ }
+ *data_size = (uint8_t*) o - (uint8_t*) data;
+ }
+
+ return buf - avpkt->data;
+}
+
+
+AVCodec ff_s302m_decoder = {
+ .name = "s302m",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_S302M,
+ .priv_data_size = 0,
+ .decode = s302m_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("SMPTE 302M"),
+};