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@@ -0,0 +1,177 @@
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+/*
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+ * RTP AMR Depacketizer, RFC 3267
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+ * Copyright (c) 2010 Martin Storsjo
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+ *
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+ * This file is part of FFmpeg.
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+ *
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+ * FFmpeg is free software; you can redistribute it and/or
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+ * modify it under the terms of the GNU Lesser General Public
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+ * License as published by the Free Software Foundation; either
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+ * version 2.1 of the License, or (at your option) any later version.
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+ *
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+ * FFmpeg is distributed in the hope that it will be useful,
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+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
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+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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+ * Lesser General Public License for more details.
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+ *
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+ * You should have received a copy of the GNU Lesser General Public
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+ * License along with FFmpeg; if not, write to the Free Software
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+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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+ */
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+
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+#include "avformat.h"
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+#include "rtpdec_amr.h"
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+#include "libavutil/avstring.h"
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+
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+static const uint8_t frame_sizes_nb[16] = {
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+ 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0
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+};
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+static const uint8_t frame_sizes_wb[16] = {
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+ 17, 23, 32, 36, 40, 46, 50, 58, 60, 5, 5, 0, 0, 0, 0, 0
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+};
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+
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+static int amr_handle_packet(AVFormatContext *ctx,
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+ PayloadContext *data,
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+ AVStream *st,
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+ AVPacket * pkt,
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+ uint32_t * timestamp,
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+ const uint8_t * buf,
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+ int len, int flags)
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+{
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+ const uint8_t *frame_sizes = NULL;
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+ int frames;
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+ int i;
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+ const uint8_t *speech_data;
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+ uint8_t *ptr;
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+
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+ if (st->codec->codec_id == CODEC_ID_AMR_NB) {
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+ frame_sizes = frame_sizes_nb;
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+ } else if (st->codec->codec_id == CODEC_ID_AMR_WB) {
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+ frame_sizes = frame_sizes_wb;
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+ } else {
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+ av_log(ctx, AV_LOG_ERROR, "Bad codec ID\n");
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+ return AVERROR_INVALIDDATA;
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+ }
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+
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+ if (st->codec->channels != 1) {
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+ av_log(ctx, AV_LOG_ERROR, "Only mono AMR is supported\n");
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+ return AVERROR_INVALIDDATA;
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+ }
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+
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+ /* The AMR RTP packet consists of one header byte, followed
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+ * by one TOC byte for each AMR frame in the packet, followed
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+ * by the speech data for all the AMR frames.
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+ *
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+ * The header byte contains only a codec mode request, for
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+ * requesting what kind of AMR data the sender wants to
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+ * receive. Not used at the moment.
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+ */
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+
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+ /* Count the number of frames in the packet. The highest bit
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+ * is set in a TOC byte if there are more frames following.
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+ */
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+ for (frames = 1; frames < len && (buf[frames] & 0x80); frames++) ;
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+
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+ if (1 + frames >= len) {
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+ /* We hit the end of the packet while counting frames. */
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+ av_log(ctx, AV_LOG_ERROR, "No speech data found\n");
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+ return AVERROR_INVALIDDATA;
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+ }
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+
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+ speech_data = buf + 1 + frames;
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+
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+ /* Everything except the codec mode request byte should be output. */
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+ if (av_new_packet(pkt, len - 1)) {
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+ av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
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+ return AVERROR_NOMEM;
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+ }
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+ pkt->stream_index = st->index;
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+ ptr = pkt->data;
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+
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+ for (i = 0; i < frames; i++) {
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+ uint8_t toc = buf[1 + i];
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+ int frame_size = frame_sizes[(toc >> 3) & 0x0f];
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+
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+ if (speech_data + frame_size > buf + len) {
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+ /* Too little speech data */
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+ av_log(ctx, AV_LOG_WARNING, "Too little speech data in the RTP packet\n");
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+ /* Set the unwritten part of the packet to zero. */
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+ memset(ptr, 0, pkt->data + pkt->size - ptr);
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+ pkt->size = ptr - pkt->data;
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+ return 0;
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+ }
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+
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+ /* Extract the AMR frame mode from the TOC byte */
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+ *ptr++ = toc & 0x7C;
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+
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+ /* Copy the speech data */
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+ memcpy(ptr, speech_data, frame_size);
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+ speech_data += frame_size;
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+ ptr += frame_size;
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+ }
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+
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+ if (speech_data < buf + len) {
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+ av_log(ctx, AV_LOG_WARNING, "Too much speech data in the RTP packet?\n");
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+ /* Set the unwritten part of the packet to zero. */
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+ memset(ptr, 0, pkt->data + pkt->size - ptr);
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+ pkt->size = ptr - pkt->data;
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+ }
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+
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+ return 0;
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+}
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+
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+static int amr_parse_sdp_line(AVFormatContext *s, int st_index,
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+ PayloadContext *data, const char *line)
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+{
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+ const char *p;
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+ char attr[25], value[25];
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+
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+ /* Parse an fmtp line this one:
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+ * a=fmtp:97 octet-align=1; interleaving=0
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+ * That is, a normal fmtp: line followed by semicolon & space
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+ * separated key/value pairs.
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+ */
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+ if (av_strstart(line, "fmtp:", &p)) {
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+ int octet_align = 0;
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+ int crc = 0;
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+ int interleaving = 0;
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+ int channels = 1;
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+
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+ while (*p && *p == ' ') p++; /* strip spaces */
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+ while (*p && *p != ' ') p++; /* eat protocol identifier */
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+ while (*p && *p == ' ') p++; /* strip trailing spaces */
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+
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+ while (rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value))) {
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+ if (!strcmp(attr, "octet-align"))
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+ octet_align = atoi(value);
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+ else if (!strcmp(attr, "crc"))
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+ crc = atoi(value);
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+ else if (!strcmp(attr, "interleaving"))
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+ interleaving = atoi(value);
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+ else if (!strcmp(attr, "channels"))
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+ channels = atoi(value);
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+ }
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+ if (!octet_align || crc || interleaving || channels != 1) {
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+ av_log(s, AV_LOG_ERROR, "Unsupported RTP/AMR configuration!\n");
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+ return -1;
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+ }
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+ }
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+ return 0;
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+}
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+
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+RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler = {
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+ .enc_name = "AMR",
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+ .codec_type = CODEC_TYPE_AUDIO,
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+ .codec_id = CODEC_ID_AMR_NB,
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+ .parse_sdp_a_line = amr_parse_sdp_line,
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+ .parse_packet = amr_handle_packet,
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+};
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+
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+RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler = {
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+ .enc_name = "AMR-WB",
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+ .codec_type = CODEC_TYPE_AUDIO,
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+ .codec_id = CODEC_ID_AMR_WB,
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+ .parse_sdp_a_line = amr_parse_sdp_line,
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+ .parse_packet = amr_handle_packet,
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+};
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+
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Ronald S. Bultje