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Diffstat (limited to 'ffmpeg1/libavformat/rtpdec_amr.c')
-rw-r--r--ffmpeg1/libavformat/rtpdec_amr.c209
1 files changed, 209 insertions, 0 deletions
diff --git a/ffmpeg1/libavformat/rtpdec_amr.c b/ffmpeg1/libavformat/rtpdec_amr.c
new file mode 100644
index 0000000..211cf17
--- /dev/null
+++ b/ffmpeg1/libavformat/rtpdec_amr.c
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+/*
+ * 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 "libavutil/channel_layout.h"
+#include "avformat.h"
+#include "rtpdec_formats.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
+};
+
+struct PayloadContext {
+ int octet_align;
+ int crc;
+ int interleaving;
+ int channels;
+};
+
+static PayloadContext *amr_new_context(void)
+{
+ PayloadContext *data = av_mallocz(sizeof(PayloadContext));
+ if(!data) return data;
+ data->channels = 1;
+ return data;
+}
+
+static void amr_free_context(PayloadContext *data)
+{
+ av_free(data);
+}
+
+static int amr_handle_packet(AVFormatContext *ctx, PayloadContext *data,
+ AVStream *st, AVPacket *pkt, uint32_t *timestamp,
+ const uint8_t *buf, int len, uint16_t seq,
+ 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 == AV_CODEC_ID_AMR_NB) {
+ frame_sizes = frame_sizes_nb;
+ } else if (st->codec->codec_id == AV_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;
+ }
+ st->codec->channel_layout = AV_CH_LAYOUT_MONO;
+
+ /* 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(ENOMEM);
+ }
+ 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_fmtp(AVStream *stream, PayloadContext *data,
+ char *attr, char *value)
+{
+ /* Some AMR SDP configurations contain "octet-align", without
+ * the trailing =1. Therefore, if the value is empty,
+ * interpret it as "1".
+ */
+ if (!strcmp(value, "")) {
+ av_log(NULL, AV_LOG_WARNING, "AMR fmtp attribute %s had "
+ "nonstandard empty value\n", attr);
+ strcpy(value, "1");
+ }
+ if (!strcmp(attr, "octet-align"))
+ data->octet_align = atoi(value);
+ else if (!strcmp(attr, "crc"))
+ data->crc = atoi(value);
+ else if (!strcmp(attr, "interleaving"))
+ data->interleaving = atoi(value);
+ else if (!strcmp(attr, "channels"))
+ data->channels = atoi(value);
+ return 0;
+}
+
+static int amr_parse_sdp_line(AVFormatContext *s, int st_index,
+ PayloadContext *data, const char *line)
+{
+ const char *p;
+ int ret;
+
+ if (st_index < 0)
+ return 0;
+
+ /* 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)) {
+ ret = ff_parse_fmtp(s->streams[st_index], data, p, amr_parse_fmtp);
+ if (!data->octet_align || data->crc ||
+ data->interleaving || data->channels != 1) {
+ av_log(s, AV_LOG_ERROR, "Unsupported RTP/AMR configuration!\n");
+ return -1;
+ }
+ return ret;
+ }
+ return 0;
+}
+
+RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler = {
+ .enc_name = "AMR",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_AMR_NB,
+ .parse_sdp_a_line = amr_parse_sdp_line,
+ .alloc = amr_new_context,
+ .free = amr_free_context,
+ .parse_packet = amr_handle_packet,
+};
+
+RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler = {
+ .enc_name = "AMR-WB",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_AMR_WB,
+ .parse_sdp_a_line = amr_parse_sdp_line,
+ .alloc = amr_new_context,
+ .free = amr_free_context,
+ .parse_packet = amr_handle_packet,
+};