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FFmpeg/libavformat/rtpdec_qcelp.c
Michael Niedermayer 7b376b398a Merge remote branch 'qatar/master'
* qatar/master:
  Handle unicode file names on windows
  rtp: Rename the open/close functions to alloc/free
  Lowercase all ff* program names.
  Refer to ff* tools by their lowercase names.
NOT Pulled  Replace more FFmpeg instances by Libav or ffmpeg.
  Replace `` by $() syntax in shell scripts.
  patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled  Remove stray libavcore and _g binary references.
  vorbis: Rename decoder/encoder files to follow general file naming scheme.
  aacenc: Fix whitespace after last commit.
  cook: Fix small typo in av_log_ask_for_sample message.
  aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
  Remove RDFT dependency from AAC decoder.
  Add some debug log messages to AAC extradata
  Fix mov debug (u)int64_t format strings.
  bswap: use native types for av_bwap16().
  doc: FLV muxing is supported.
  applehttp: Handle AES-128 encrypted streams
  Add a protocol handler for AES CBC decryption with PKCS7 padding
  doc: Mention that DragonFly BSD requires __BSD_VISIBLE set

Conflicts:
	ffplay.c
	ffprobe.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-24 03:41:22 +02:00

230 lines
7.7 KiB
C

/**
* RTP Depacketization of QCELP/PureVoice, RFC 2658
* 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 "rtpdec_formats.h"
static const uint8_t frame_sizes[] = {
1, 4, 8, 17, 35
};
typedef struct {
int pos;
int size;
/* The largest frame is 35 bytes, only 10 frames are allowed per
* packet, and we return the first one immediately, so allocate
* space for 9 frames */
uint8_t data[35*9];
} InterleavePacket;
struct PayloadContext {
int interleave_size;
int interleave_index;
InterleavePacket group[6];
int group_finished;
/* The maximum packet size, 10 frames of 35 bytes each, and one
* packet header byte. */
uint8_t next_data[1 + 35*10];
int next_size;
uint32_t next_timestamp;
};
static PayloadContext *qcelp_new_context(void)
{
return av_mallocz(sizeof(PayloadContext));
}
static void qcelp_free_context(PayloadContext *data)
{
av_free(data);
}
static int return_stored_frame(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len);
static int store_packet(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len)
{
int interleave_size, interleave_index;
int frame_size, ret;
InterleavePacket* ip;
if (len < 2)
return AVERROR_INVALIDDATA;
interleave_size = buf[0] >> 3 & 7;
interleave_index = buf[0] & 7;
if (interleave_size > 5) {
av_log(ctx, AV_LOG_ERROR, "Invalid interleave size %d\n",
interleave_size);
return AVERROR_INVALIDDATA;
}
if (interleave_index > interleave_size) {
av_log(ctx, AV_LOG_ERROR, "Invalid interleave index %d/%d\n",
interleave_index, interleave_size);
return AVERROR_INVALIDDATA;
}
if (interleave_size != data->interleave_size) {
int i;
/* First packet, or changed interleave size */
data->interleave_size = interleave_size;
data->interleave_index = 0;
for (i = 0; i < 6; i++)
data->group[i].size = 0;
}
if (interleave_index < data->interleave_index) {
/* Wrapped around - missed the last packet of the previous group. */
if (data->group_finished) {
/* No more data in the packets in this interleaving group, just
* start processing the next one */
data->interleave_index = 0;
} else {
/* Stash away the current packet, emit everything we have of the
* previous group. */
for (; data->interleave_index <= interleave_size;
data->interleave_index++)
data->group[data->interleave_index].size = 0;
if (len > sizeof(data->next_data))
return AVERROR_INVALIDDATA;
memcpy(data->next_data, buf, len);
data->next_size = len;
data->next_timestamp = *timestamp;
*timestamp = RTP_NOTS_VALUE;
data->interleave_index = 0;
return return_stored_frame(ctx, data, st, pkt, timestamp, buf, len);
}
}
if (interleave_index > data->interleave_index) {
/* We missed a packet */
for (; data->interleave_index < interleave_index;
data->interleave_index++)
data->group[data->interleave_index].size = 0;
}
data->interleave_index = interleave_index;
if (buf[1] >= FF_ARRAY_ELEMS(frame_sizes))
return AVERROR_INVALIDDATA;
frame_size = frame_sizes[buf[1]];
if (1 + frame_size > len)
return AVERROR_INVALIDDATA;
if (len - 1 - frame_size > sizeof(data->group[0].data))
return AVERROR_INVALIDDATA;
if ((ret = av_new_packet(pkt, frame_size)) < 0)
return ret;
memcpy(pkt->data, &buf[1], frame_size);
pkt->stream_index = st->index;
ip = &data->group[data->interleave_index];
ip->size = len - 1 - frame_size;
ip->pos = 0;
memcpy(ip->data, &buf[1 + frame_size], ip->size);
/* Each packet must contain the same number of frames according to the
* RFC. If there's no data left in this packet, there shouldn't be any
* in any of the other frames in the interleaving group either. */
data->group_finished = ip->size == 0;
if (interleave_index == interleave_size) {
data->interleave_index = 0;
return !data->group_finished;
} else {
data->interleave_index++;
return 0;
}
}
static int return_stored_frame(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len)
{
InterleavePacket* ip = &data->group[data->interleave_index];
int frame_size, ret;
if (data->group_finished && data->interleave_index == 0) {
*timestamp = data->next_timestamp;
ret = store_packet(ctx, data, st, pkt, timestamp, data->next_data,
data->next_size);
data->next_size = 0;
return ret;
}
if (ip->size == 0) {
/* No stored data for this interleave block, output an empty packet */
if ((ret = av_new_packet(pkt, 1)) < 0)
return ret;
pkt->data[0] = 0; // Blank - could also be 14, Erasure
} else {
if (ip->pos >= ip->size)
return AVERROR_INVALIDDATA;
if (ip->data[ip->pos] >= FF_ARRAY_ELEMS(frame_sizes))
return AVERROR_INVALIDDATA;
frame_size = frame_sizes[ip->data[ip->pos]];
if (ip->pos + frame_size > ip->size)
return AVERROR_INVALIDDATA;
if ((ret = av_new_packet(pkt, frame_size)) < 0)
return ret;
memcpy(pkt->data, &ip->data[ip->pos], frame_size);
ip->pos += frame_size;
data->group_finished = ip->pos >= ip->size;
}
pkt->stream_index = st->index;
if (data->interleave_index == data->interleave_size) {
data->interleave_index = 0;
if (!data->group_finished)
return 1;
else
return data->next_size > 0;
} else {
data->interleave_index++;
return 1;
}
}
static int qcelp_parse_packet(AVFormatContext *ctx, PayloadContext *data,
AVStream *st, AVPacket *pkt, uint32_t *timestamp,
const uint8_t *buf, int len, int flags)
{
if (buf)
return store_packet(ctx, data, st, pkt, timestamp, buf, len);
else
return return_stored_frame(ctx, data, st, pkt, timestamp, buf, len);
}
RTPDynamicProtocolHandler ff_qcelp_dynamic_handler = {
.enc_name = "x-Purevoice",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = CODEC_ID_QCELP,
.static_payload_id = 12,
.alloc = qcelp_new_context,
.free = qcelp_free_context,
.parse_packet = qcelp_parse_packet
};