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ffmpeg / libavformat / rtpdec_amr.c @ 9261e6cf

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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 Libav.
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 *
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 * Libav 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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 * Libav 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 Libav; 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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#include "avformat.h"
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#include "rtpdec_formats.h"
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#include "libavutil/avstring.h"
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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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struct PayloadContext {
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    int octet_align;
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    int crc;
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    int interleaving;
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    int channels;
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};
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static PayloadContext *amr_new_context(void)
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{
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    PayloadContext *data = av_mallocz(sizeof(PayloadContext));
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    if(!data) return data;
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    data->channels = 1;
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    return data;
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}
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static void amr_free_context(PayloadContext *data)
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{
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    av_free(data);
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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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    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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    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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    /* 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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    /* 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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    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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    speech_data = buf + 1 + frames;
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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(ENOMEM);
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    }
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    pkt->stream_index = st->index;
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    ptr = pkt->data;
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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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        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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        /* Extract the AMR frame mode from the TOC byte */
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        *ptr++ = toc & 0x7C;
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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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    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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    return 0;
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}
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static int amr_parse_fmtp(AVStream *stream, PayloadContext *data,
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                          char *attr, char *value)
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{
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    /* Some AMR SDP configurations contain "octet-align", without
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     * the trailing =1. Therefore, if the value is empty,
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     * interpret it as "1".
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     */
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    if (!strcmp(value, "")) {
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        av_log(NULL, AV_LOG_WARNING, "AMR fmtp attribute %s had "
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                                     "nonstandard empty value\n", attr);
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        strcpy(value, "1");
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    }
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    if (!strcmp(attr, "octet-align"))
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        data->octet_align = atoi(value);
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    else if (!strcmp(attr, "crc"))
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        data->crc = atoi(value);
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    else if (!strcmp(attr, "interleaving"))
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        data->interleaving = atoi(value);
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    else if (!strcmp(attr, "channels"))
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        data->channels = atoi(value);
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    return 0;
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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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    int ret;
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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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        ret = ff_parse_fmtp(s->streams[st_index], data, p, amr_parse_fmtp);
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        if (!data->octet_align || data->crc ||
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            data->interleaving || data->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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        return ret;
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    }
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    return 0;
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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       = AVMEDIA_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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    .alloc            = amr_new_context,
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    .free             = amr_free_context,
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    .parse_packet     = amr_handle_packet,
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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       = AVMEDIA_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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    .alloc            = amr_new_context,
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    .free             = amr_free_context,
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    .parse_packet     = amr_handle_packet,
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};
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