1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavformat/rtpdec_amr.c
Andreas Rheinhardt c1e439d7e9 avformat: Forward errors where possible
It is not uncommon to find code where the caller thinks to know better
what the return value should be than the callee. E.g. something like
"if (av_new_packet(pkt, size) < 0) return AVERROR(ENOMEM);". This commit
changes several instances of this to instead forward the actual error.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2019-12-12 19:25:33 +01:00

204 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 "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 av_cold int amr_init(AVFormatContext *s, int st_index, PayloadContext *data)
{
data->channels = 1;
return 0;
}
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, ret;
const uint8_t *speech_data;
uint8_t *ptr;
if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB) {
frame_sizes = frame_sizes_nb;
} else if (st->codecpar->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->codecpar->channels != 1) {
av_log(ctx, AV_LOG_ERROR, "Only mono AMR is supported\n");
return AVERROR_INVALIDDATA;
}
st->codecpar->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 ((ret = av_new_packet(pkt, len - 1)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Out of memory\n");
return ret;
}
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(AVFormatContext *s,
AVStream *stream, PayloadContext *data,
const char *attr, const 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(s, AV_LOG_WARNING, "AMR fmtp attribute %s had "
"nonstandard empty value\n", attr);
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, 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;
}
const RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler = {
.enc_name = "AMR",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_AMR_NB,
.priv_data_size = sizeof(PayloadContext),
.init = amr_init,
.parse_sdp_a_line = amr_parse_sdp_line,
.parse_packet = amr_handle_packet,
};
const RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler = {
.enc_name = "AMR-WB",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_AMR_WB,
.priv_data_size = sizeof(PayloadContext),
.init = amr_init,
.parse_sdp_a_line = amr_parse_sdp_line,
.parse_packet = amr_handle_packet,
};