1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavformat/rtpdec_amr.c
Michael Niedermayer c065255bba Merge remote-tracking branch 'qatar/master'
* qatar/master:
  aacenc: Fix LONG_START windowing.
  aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
  avplay: use the correct array size for stride.
  lavc: extend doxy for avcodec_alloc_context3().
  APIchanges: mention avcodec_alloc_context()/2/3
  avcodec_align_dimensions2: set only 4 linesizes, not AV_NUM_DATA_POINTERS.
  aacsbr: ARM NEON optimised sbrdsp functions
  aacsbr: align some arrays
  aacsbr: move some simdable loops to function pointers
  cosmetics: Remove extra newlines at EOF

Conflicts:
	libavcodec/utils.c
	libavfilter/formats.c
	libavutil/mem.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-29 00:34:59 +01:00

208 lines
6.6 KiB
C

/*
* 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_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, 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(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;
/* 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 = 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 = 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,
};