ffmpeg roll of source to mar 9 version...

patched-ffmpeg-mt/libavformat/rtpdec_amr.c

Index: patched-ffmpeg-mt/libavformat/rtpdec_amr.c
===================================================================
--- patched-ffmpeg-mt/libavformat/rtpdec_amr.c	(revision 0)
+++ patched-ffmpeg-mt/libavformat/rtpdec_amr.c	(revision 0)
@@ -0,0 +1,177 @@
+/*
+ * RTP AMR Depacketizer, RFC 3267
+ * 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 "avformat.h"
+#include "rtpdec_amr.h"
+#include "libavutil/avstring.h"
+
+static const uint8_t frame_sizes_nb[16] = {
+    12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0
+};
+static const uint8_t frame_sizes_wb[16] = {
+    17, 23, 32, 36, 40, 46, 50, 58, 60, 5, 5, 0, 0, 0, 0, 0
+};
+
+static int amr_handle_packet(AVFormatContext *ctx,
+                             PayloadContext *data,
+                             AVStream *st,
+                             AVPacket * pkt,
+                             uint32_t * timestamp,
+                             const uint8_t * buf,
+                             int len, int flags)
+{
+    const uint8_t *frame_sizes = NULL;
+    int frames;
+    int i;
+    const uint8_t *speech_data;
+    uint8_t *ptr;
+
+    if (st->codec->codec_id == CODEC_ID_AMR_NB) {
+        frame_sizes = frame_sizes_nb;
+    } else if (st->codec->codec_id == CODEC_ID_AMR_WB) {
+        frame_sizes = frame_sizes_wb;
+    } else {
+        av_log(ctx, AV_LOG_ERROR, "Bad codec ID\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (st->codec->channels != 1) {
+        av_log(ctx, AV_LOG_ERROR, "Only mono AMR is supported\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* The AMR RTP packet consists of one header byte, followed
+     * by one TOC byte for each AMR frame in the packet, followed
+     * by the speech data for all the AMR frames.
+     *
+     * The header byte contains only a codec mode request, for
+     * requesting what kind of AMR data the sender wants to
+     * receive. Not used at the moment.
+     */
+
+    /* Count the number of frames in the packet. The highest bit
+     * is set in a TOC byte if there are more frames following.
+     */
+    for (frames = 1; frames < len && (buf[frames] & 0x80); frames++) ;
+
+    if (1 + frames >= len) {
+        /* We hit the end of the packet while counting frames. */
+        av_log(ctx, AV_LOG_ERROR, "No speech data found\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    speech_data = buf + 1 + frames;
+
+    /* Everything except the codec mode request byte should be output. */
+    if (av_new_packet(pkt, len - 1)) {
+        av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
+        return AVERROR_NOMEM;
+    }
+    pkt->stream_index = st->index;
+    ptr = pkt->data;
+
+    for (i = 0; i < frames; i++) {
+        uint8_t toc = buf[1 + i];
+        int frame_size = frame_sizes[(toc >> 3) & 0x0f];
+
+        if (speech_data + frame_size > buf + len) {
+            /* Too little speech data */
+            av_log(ctx, AV_LOG_WARNING, "Too little speech data in the RTP packet\n");
+            /* Set the unwritten part of the packet to zero. */
+            memset(ptr, 0, pkt->data + pkt->size - ptr);
+            pkt->size = ptr - pkt->data;
+            return 0;
+        }
+
+        /* Extract the AMR frame mode from the TOC byte */
+        *ptr++ = toc & 0x7C;
+
+        /* Copy the speech data */
+        memcpy(ptr, speech_data, frame_size);
+        speech_data += frame_size;
+        ptr += frame_size;
+    }
+
+    if (speech_data < buf + len) {
+        av_log(ctx, AV_LOG_WARNING, "Too much speech data in the RTP packet?\n");
+        /* Set the unwritten part of the packet to zero. */
+        memset(ptr, 0, pkt->data + pkt->size - ptr);
+        pkt->size = ptr - pkt->data;
+    }
+
+    return 0;
+}
+
+static int amr_parse_sdp_line(AVFormatContext *s, int st_index,
+                              PayloadContext *data, const char *line)
+{
+    const char *p;
+    char attr[25], value[25];
+
+    /* Parse an fmtp line this one:
+     * a=fmtp:97 octet-align=1; interleaving=0
+     * That is, a normal fmtp: line followed by semicolon & space
+     * separated key/value pairs.
+     */
+    if (av_strstart(line, "fmtp:", &p)) {
+        int octet_align = 0;
+        int crc = 0;
+        int interleaving = 0;
+        int channels = 1;
+
+        while (*p && *p == ' ') p++; /* strip spaces */
+        while (*p && *p != ' ') p++; /* eat protocol identifier */
+        while (*p && *p == ' ') p++; /* strip trailing spaces */
+
+        while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value))) {
+            if (!strcmp(attr, "octet-align"))
+                octet_align = atoi(value);
+            else if (!strcmp(attr, "crc"))
+                crc = atoi(value);
+            else if (!strcmp(attr, "interleaving"))
+                interleaving = atoi(value);
+            else if (!strcmp(attr, "channels"))
+                channels = atoi(value);
+        }
+        if (!octet_align || crc || interleaving || channels != 1) {
+            av_log(s, AV_LOG_ERROR, "Unsupported RTP/AMR configuration!\n");
+            return -1;
+        }
+    }
+    return 0;
+}
+
+RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler = {
+    .enc_name         = "AMR",
+    .codec_type       = CODEC_TYPE_AUDIO,
+    .codec_id         = CODEC_ID_AMR_NB,
+    .parse_sdp_a_line = amr_parse_sdp_line,
+    .parse_packet     = amr_handle_packet,
+};
+
+RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler = {
+    .enc_name         = "AMR-WB",
+    .codec_type       = CODEC_TYPE_AUDIO,
+    .codec_id         = CODEC_ID_AMR_WB,
+    .parse_sdp_a_line = amr_parse_sdp_line,
+    .parse_packet     = amr_handle_packet,
+};
+