ffmpeg source update for sep 09

source/patched-ffmpeg-mt/libavformat/rtpdec_latm.c

Index: source/patched-ffmpeg-mt/libavformat/rtpdec_latm.c
===================================================================
--- source/patched-ffmpeg-mt/libavformat/rtpdec_latm.c	(revision 0)
+++ source/patched-ffmpeg-mt/libavformat/rtpdec_latm.c	(revision 0)
@@ -0,0 +1,187 @@
+/**
+ * RTP Depacketization of MP4A-LATM, RFC 3016
+ * Copyright (c) 2010 Martin Storsjo
+ *
+ * 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 "rtpdec_formats.h"
+#include "internal.h"
+#include "libavutil/avstring.h"
+#include "libavcodec/get_bits.h"
+#include <strings.h>
+
+struct PayloadContext {
+    ByteIOContext *dyn_buf;
+    uint8_t *buf;
+    int pos, len;
+    uint32_t timestamp;
+};
+
+static PayloadContext *latm_new_context(void)
+{
+    return av_mallocz(sizeof(PayloadContext));
+}
+
+static void latm_free_context(PayloadContext *data)
+{
+    if (!data)
+        return;
+    if (data->dyn_buf) {
+        uint8_t *p;
+        url_close_dyn_buf(data->dyn_buf, &p);
+        av_free(p);
+    }
+    av_free(data->buf);
+    av_free(data);
+}
+
+static int latm_parse_packet(AVFormatContext *ctx, PayloadContext *data,
+                             AVStream *st, AVPacket *pkt, uint32_t *timestamp,
+                             const uint8_t *buf, int len, int flags)
+{
+    int ret, cur_len;
+
+    if (buf) {
+        if (!data->dyn_buf || data->timestamp != *timestamp) {
+            av_freep(&data->buf);
+            if (data->dyn_buf)
+                url_close_dyn_buf(data->dyn_buf, &data->buf);
+            data->dyn_buf = NULL;
+            av_freep(&data->buf);
+
+            data->timestamp = *timestamp;
+            if ((ret = url_open_dyn_buf(&data->dyn_buf)) < 0)
+                return ret;
+        }
+        put_buffer(data->dyn_buf, buf, len);
+
+        if (!(flags & RTP_FLAG_MARKER))
+            return AVERROR(EAGAIN);
+        av_free(data->buf);
+        data->len = url_close_dyn_buf(data->dyn_buf, &data->buf);
+        data->dyn_buf = NULL;
+        data->pos = 0;
+    }
+
+    if (!data->buf) {
+        av_log(ctx, AV_LOG_ERROR, "No data available yet\n");
+        return AVERROR(EIO);
+    }
+
+    cur_len = 0;
+    while (data->pos < data->len) {
+        uint8_t val = data->buf[data->pos++];
+        cur_len += val;
+        if (val != 0xff)
+            break;
+    }
+    if (data->pos + cur_len > data->len) {
+        av_log(ctx, AV_LOG_ERROR, "Malformed LATM packet\n");
+        return AVERROR(EIO);
+    }
+
+    if ((ret = av_new_packet(pkt, cur_len)) < 0)
+        return ret;
+    memcpy(pkt->data, data->buf + data->pos, cur_len);
+    data->pos += cur_len;
+    pkt->stream_index = st->index;
+    return data->pos < data->len;
+}
+
+static int parse_fmtp_config(AVStream *st, char *value)
+{
+    int len = ff_hex_to_data(NULL, value), i, ret = 0;
+    GetBitContext gb;
+    uint8_t *config;
+    int audio_mux_version, same_time_framing, num_sub_frames,
+        num_programs, num_layers;
+
+    /* Pad this buffer, too, to avoid out of bounds reads with get_bits below */
+    config = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+    if (!config)
+        return AVERROR(ENOMEM);
+    ff_hex_to_data(config, value);
+    init_get_bits(&gb, config, len*8);
+    audio_mux_version = get_bits(&gb, 1);
+    same_time_framing = get_bits(&gb, 1);
+    num_sub_frames    = get_bits(&gb, 6);
+    num_programs      = get_bits(&gb, 4);
+    num_layers        = get_bits(&gb, 3);
+    if (audio_mux_version != 0 || same_time_framing != 1 || num_programs != 0 ||
+        num_layers != 0) {
+        av_log(NULL, AV_LOG_WARNING, "Unsupported LATM config (%d,%d,%d,%d)\n",
+                                     audio_mux_version, same_time_framing,
+                                     num_programs, num_layers);
+        ret = AVERROR_PATCHWELCOME;
+        goto end;
+    }
+    av_freep(&st->codec->extradata);
+    st->codec->extradata_size = (get_bits_left(&gb) + 7)/8;
+    st->codec->extradata = av_mallocz(st->codec->extradata_size +
+                                      FF_INPUT_BUFFER_PADDING_SIZE);
+    if (!st->codec->extradata) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    for (i = 0; i < st->codec->extradata_size; i++)
+        st->codec->extradata[i] = get_bits(&gb, 8);
+
+end:
+    av_free(config);
+    return ret;
+}
+
+static int parse_fmtp(AVStream *stream, PayloadContext *data,
+                      char *attr, char *value)
+{
+    int res;
+
+    if (!strcmp(attr, "config")) {
+        res = parse_fmtp_config(stream, value);
+        if (res < 0)
+            return res;
+    } else if (!strcmp(attr, "cpresent")) {
+        int cpresent = atoi(value);
+        if (cpresent != 0)
+            av_log_missing_feature(NULL, "RTP MP4A-LATM with in-band "
+                                         "configuration", 1);
+    }
+
+    return 0;
+}
+
+static int latm_parse_sdp_line(AVFormatContext *s, int st_index,
+                               PayloadContext *data, const char *line)
+{
+    const char *p;
+
+    if (av_strstart(line, "fmtp:", &p))
+        return ff_parse_fmtp(s->streams[st_index], data, p, parse_fmtp);
+
+    return 0;
+}
+
+RTPDynamicProtocolHandler ff_mp4a_latm_dynamic_handler = {
+    .enc_name           = "MP4A-LATM",
+    .codec_type         = AVMEDIA_TYPE_AUDIO,
+    .codec_id           = CODEC_ID_AAC,
+    .parse_sdp_a_line   = latm_parse_sdp_line,
+    .open               = latm_new_context,
+    .close              = latm_free_context,
+    .parse_packet       = latm_parse_packet
+};