Mercurial > libavformat.hg
view flvdec.c @ 4723:a2390c6a35e6 libavformat
Fix index generation in the way that it was supposed to be used. See the
discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading
code".
Over time, this code broke somewhat, e.g. seq was never actually written
into (and was thus always 1, therefore the seq condition was always true),
whereas it was supposed to be set to the sequence number of the video slice
in case the video frame is divided over multiple RM packets (slices). The
problem of this is that packets other than those containing the beginning
of a video frame would be indexed as well.
Secondly, flags&2 is supposed to be true for video keyframes and for these
audio packets containing the start of a block. For some codecs (e.g. AAC),
that is every single packet, whereas for others (e.g. cook), that is the
packet containing the first of a series of scrambled packets that are to be
descrambled together. Indexing any of the following would lead to incomplete
and thus useless frames. Problem here is that flags would be reset to 2 to
indicate that the first packet is ready to be returned, and in addition if
no data was left to be returned (which is always true for the first packet),
then we wouldn't actually write the index entry anyway.
All in all, the idea was good and it probably worked at some point, but that
is long ago. This patch should at the very least make it likely for this code
to be executed again at the right times, i.e. the way it was originally
intended to be used.
author | rbultje |
---|---|
date | Sun, 15 Mar 2009 20:14:25 +0000 |
parents | d148dbaebaca |
children | 063522a177cc |
line wrap: on
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/* * FLV demuxer * Copyright (c) 2003 The FFmpeg Project * * This demuxer will generate a 1 byte extradata for VP6F content. * It is composed of: * - upper 4bits: difference between encoded width and visible width * - lower 4bits: difference between encoded height and visible height * * 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 "libavcodec/mpeg4audio.h" #include "avformat.h" #include "flv.h" typedef struct { int wrong_dts; ///< wrong dts due to negative cts } FLVContext; static int flv_probe(AVProbeData *p) { const uint8_t *d; d = p->buf; if (d[0] == 'F' && d[1] == 'L' && d[2] == 'V' && d[3] < 5 && d[5]==0) { return AVPROBE_SCORE_MAX; } return 0; } static void flv_set_audio_codec(AVFormatContext *s, AVStream *astream, int flv_codecid) { AVCodecContext *acodec = astream->codec; switch(flv_codecid) { //no distinction between S16 and S8 PCM codec flags case FLV_CODECID_PCM: acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : #ifdef WORDS_BIGENDIAN CODEC_ID_PCM_S16BE; #else CODEC_ID_PCM_S16LE; #endif break; case FLV_CODECID_PCM_LE: acodec->codec_id = acodec->bits_per_coded_sample == 8 ? CODEC_ID_PCM_S8 : CODEC_ID_PCM_S16LE; break; case FLV_CODECID_AAC : acodec->codec_id = CODEC_ID_AAC; break; case FLV_CODECID_ADPCM: acodec->codec_id = CODEC_ID_ADPCM_SWF; break; case FLV_CODECID_SPEEX: acodec->codec_id = CODEC_ID_SPEEX; acodec->sample_rate = 16000; break; case FLV_CODECID_MP3 : acodec->codec_id = CODEC_ID_MP3 ; astream->need_parsing = AVSTREAM_PARSE_FULL; break; case FLV_CODECID_NELLYMOSER_8KHZ_MONO: acodec->sample_rate = 8000; //in case metadata does not otherwise declare samplerate case FLV_CODECID_NELLYMOSER: acodec->codec_id = CODEC_ID_NELLYMOSER; break; default: av_log(s, AV_LOG_INFO, "Unsupported audio codec (%x)\n", flv_codecid >> FLV_AUDIO_CODECID_OFFSET); acodec->codec_tag = flv_codecid >> FLV_AUDIO_CODECID_OFFSET; } } static int flv_set_video_codec(AVFormatContext *s, AVStream *vstream, int flv_codecid) { AVCodecContext *vcodec = vstream->codec; switch(flv_codecid) { case FLV_CODECID_H263 : vcodec->codec_id = CODEC_ID_FLV1 ; break; case FLV_CODECID_SCREEN: vcodec->codec_id = CODEC_ID_FLASHSV; break; case FLV_CODECID_VP6 : vcodec->codec_id = CODEC_ID_VP6F ; case FLV_CODECID_VP6A : if(flv_codecid == FLV_CODECID_VP6A) vcodec->codec_id = CODEC_ID_VP6A; if(vcodec->extradata_size != 1) { vcodec->extradata_size = 1; vcodec->extradata = av_malloc(1); } vcodec->extradata[0] = get_byte(s->pb); return 1; // 1 byte body size adjustment for flv_read_packet() case FLV_CODECID_H264: vcodec->codec_id = CODEC_ID_H264; return 3; // not 4, reading packet type will consume one byte default: av_log(s, AV_LOG_INFO, "Unsupported video codec (%x)\n", flv_codecid); vcodec->codec_tag = flv_codecid; } return 0; } static int amf_get_string(ByteIOContext *ioc, char *buffer, int buffsize) { int length = get_be16(ioc); if(length >= buffsize) { url_fskip(ioc, length); return -1; } get_buffer(ioc, buffer, length); buffer[length] = '\0'; return length; } static int amf_parse_object(AVFormatContext *s, AVStream *astream, AVStream *vstream, const char *key, int64_t max_pos, int depth) { AVCodecContext *acodec, *vcodec; ByteIOContext *ioc; AMFDataType amf_type; char str_val[256]; double num_val; num_val = 0; ioc = s->pb; amf_type = get_byte(ioc); switch(amf_type) { case AMF_DATA_TYPE_NUMBER: num_val = av_int2dbl(get_be64(ioc)); break; case AMF_DATA_TYPE_BOOL: num_val = get_byte(ioc); break; case AMF_DATA_TYPE_STRING: if(amf_get_string(ioc, str_val, sizeof(str_val)) < 0) return -1; break; case AMF_DATA_TYPE_OBJECT: { unsigned int keylen; while(url_ftell(ioc) < max_pos - 2 && (keylen = get_be16(ioc))) { url_fskip(ioc, keylen); //skip key string if(amf_parse_object(s, NULL, NULL, NULL, max_pos, depth + 1) < 0) return -1; //if we couldn't skip, bomb out. } if(get_byte(ioc) != AMF_END_OF_OBJECT) return -1; } break; case AMF_DATA_TYPE_NULL: case AMF_DATA_TYPE_UNDEFINED: case AMF_DATA_TYPE_UNSUPPORTED: break; //these take up no additional space case AMF_DATA_TYPE_MIXEDARRAY: url_fskip(ioc, 4); //skip 32-bit max array index while(url_ftell(ioc) < max_pos - 2 && amf_get_string(ioc, str_val, sizeof(str_val)) > 0) { //this is the only case in which we would want a nested parse to not skip over the object if(amf_parse_object(s, astream, vstream, str_val, max_pos, depth + 1) < 0) return -1; } if(get_byte(ioc) != AMF_END_OF_OBJECT) return -1; break; case AMF_DATA_TYPE_ARRAY: { unsigned int arraylen, i; arraylen = get_be32(ioc); for(i = 0; i < arraylen && url_ftell(ioc) < max_pos - 1; i++) { if(amf_parse_object(s, NULL, NULL, NULL, max_pos, depth + 1) < 0) return -1; //if we couldn't skip, bomb out. } } break; case AMF_DATA_TYPE_DATE: url_fskip(ioc, 8 + 2); //timestamp (double) and UTC offset (int16) break; default: //unsupported type, we couldn't skip return -1; } if(depth == 1 && key) { //only look for metadata values when we are not nested and key != NULL acodec = astream ? astream->codec : NULL; vcodec = vstream ? vstream->codec : NULL; if(amf_type == AMF_DATA_TYPE_BOOL) { if(!strcmp(key, "stereo") && acodec) acodec->channels = num_val > 0 ? 2 : 1; } else if(amf_type == AMF_DATA_TYPE_NUMBER) { if(!strcmp(key, "duration")) s->duration = num_val * AV_TIME_BASE; // else if(!strcmp(key, "width") && vcodec && num_val > 0) vcodec->width = num_val; // else if(!strcmp(key, "height") && vcodec && num_val > 0) vcodec->height = num_val; else if(!strcmp(key, "videodatarate") && vcodec && 0 <= (int)(num_val * 1024.0)) vcodec->bit_rate = num_val * 1024.0; else if(!strcmp(key, "audiocodecid") && acodec && 0 <= (int)num_val) flv_set_audio_codec(s, astream, (int)num_val << FLV_AUDIO_CODECID_OFFSET); else if(!strcmp(key, "videocodecid") && vcodec && 0 <= (int)num_val) flv_set_video_codec(s, vstream, (int)num_val); else if(!strcmp(key, "audiosamplesize") && acodec && 0 < (int)num_val) { acodec->bits_per_coded_sample = num_val; //we may have to rewrite a previously read codecid because FLV only marks PCM endianness. if(num_val == 8 && (acodec->codec_id == CODEC_ID_PCM_S16BE || acodec->codec_id == CODEC_ID_PCM_S16LE)) acodec->codec_id = CODEC_ID_PCM_S8; } else if(!strcmp(key, "audiosamplerate") && acodec && num_val >= 0) { //some tools, like FLVTool2, write consistently approximate metadata sample rates if (!acodec->sample_rate) { switch((int)num_val) { case 44000: acodec->sample_rate = 44100 ; break; case 22000: acodec->sample_rate = 22050 ; break; case 11000: acodec->sample_rate = 11025 ; break; case 5000 : acodec->sample_rate = 5512 ; break; default : acodec->sample_rate = num_val; } } } } } return 0; } static int flv_read_metabody(AVFormatContext *s, int64_t next_pos) { AMFDataType type; AVStream *stream, *astream, *vstream; ByteIOContext *ioc; int i, keylen; char buffer[11]; //only needs to hold the string "onMetaData". Anything longer is something we don't want. astream = NULL; vstream = NULL; keylen = 0; ioc = s->pb; //first object needs to be "onMetaData" string type = get_byte(ioc); if(type != AMF_DATA_TYPE_STRING || amf_get_string(ioc, buffer, sizeof(buffer)) < 0 || strcmp(buffer, "onMetaData")) return -1; //find the streams now so that amf_parse_object doesn't need to do the lookup every time it is called. for(i = 0; i < s->nb_streams; i++) { stream = s->streams[i]; if (stream->codec->codec_type == CODEC_TYPE_AUDIO) astream = stream; else if(stream->codec->codec_type == CODEC_TYPE_VIDEO) vstream = stream; } //parse the second object (we want a mixed array) if(amf_parse_object(s, astream, vstream, buffer, next_pos, 0) < 0) return -1; return 0; } static AVStream *create_stream(AVFormatContext *s, int is_audio){ AVStream *st = av_new_stream(s, is_audio); if (!st) return NULL; st->codec->codec_type = is_audio ? CODEC_TYPE_AUDIO : CODEC_TYPE_VIDEO; av_set_pts_info(st, 32, 1, 1000); /* 32 bit pts in ms */ return st; } static int flv_read_header(AVFormatContext *s, AVFormatParameters *ap) { int offset, flags; url_fskip(s->pb, 4); flags = get_byte(s->pb); /* old flvtool cleared this field */ /* FIXME: better fix needed */ if (!flags) { flags = FLV_HEADER_FLAG_HASVIDEO | FLV_HEADER_FLAG_HASAUDIO; av_log(s, AV_LOG_WARNING, "Broken FLV file, which says no streams present, this might fail\n"); } if((flags & (FLV_HEADER_FLAG_HASVIDEO|FLV_HEADER_FLAG_HASAUDIO)) != (FLV_HEADER_FLAG_HASVIDEO|FLV_HEADER_FLAG_HASAUDIO)) s->ctx_flags |= AVFMTCTX_NOHEADER; if(flags & FLV_HEADER_FLAG_HASVIDEO){ if(!create_stream(s, 0)) return AVERROR(ENOMEM); } if(flags & FLV_HEADER_FLAG_HASAUDIO){ if(!create_stream(s, 1)) return AVERROR(ENOMEM); } offset = get_be32(s->pb); url_fseek(s->pb, offset, SEEK_SET); s->start_time = 0; return 0; } static int flv_get_extradata(AVFormatContext *s, AVStream *st, int size) { av_free(st->codec->extradata); st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE); if (!st->codec->extradata) return AVERROR(ENOMEM); st->codec->extradata_size = size; get_buffer(s->pb, st->codec->extradata, st->codec->extradata_size); return 0; } static int flv_read_packet(AVFormatContext *s, AVPacket *pkt) { FLVContext *flv = s->priv_data; int ret, i, type, size, flags, is_audio; int64_t next, pos; int64_t dts, pts = AV_NOPTS_VALUE; AVStream *st = NULL; for(;;){ pos = url_ftell(s->pb); url_fskip(s->pb, 4); /* size of previous packet */ type = get_byte(s->pb); size = get_be24(s->pb); dts = get_be24(s->pb); dts |= get_byte(s->pb) << 24; // av_log(s, AV_LOG_DEBUG, "type:%d, size:%d, dts:%d\n", type, size, dts); if (url_feof(s->pb)) return AVERROR_EOF; url_fskip(s->pb, 3); /* stream id, always 0 */ flags = 0; if(size == 0) continue; next= size + url_ftell(s->pb); if (type == FLV_TAG_TYPE_AUDIO) { is_audio=1; flags = get_byte(s->pb); size--; } else if (type == FLV_TAG_TYPE_VIDEO) { is_audio=0; flags = get_byte(s->pb); size--; if ((flags & 0xf0) == 0x50) /* video info / command frame */ goto skip; } else { if (type == FLV_TAG_TYPE_META && size > 13+1+4) flv_read_metabody(s, next); else /* skip packet */ av_log(s, AV_LOG_ERROR, "skipping flv packet: type %d, size %d, flags %d\n", type, size, flags); skip: url_fseek(s->pb, next, SEEK_SET); continue; } /* skip empty data packets */ if (!size) continue; /* now find stream */ for(i=0;i<s->nb_streams;i++) { st = s->streams[i]; if (st->id == is_audio) break; } if(i == s->nb_streams){ av_log(s, AV_LOG_ERROR, "invalid stream\n"); st= create_stream(s, is_audio); s->ctx_flags &= ~AVFMTCTX_NOHEADER; } // av_log(s, AV_LOG_DEBUG, "%d %X %d \n", is_audio, flags, st->discard); if( (st->discard >= AVDISCARD_NONKEY && !((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_KEY || is_audio)) ||(st->discard >= AVDISCARD_BIDIR && ((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_DISP_INTER && !is_audio)) || st->discard >= AVDISCARD_ALL ){ url_fseek(s->pb, next, SEEK_SET); continue; } if ((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_KEY) av_add_index_entry(st, pos, dts, size, 0, AVINDEX_KEYFRAME); break; } // if not streamed and no duration from metadata then seek to end to find the duration from the timestamps if(!url_is_streamed(s->pb) && s->duration==AV_NOPTS_VALUE){ int size; const int64_t pos= url_ftell(s->pb); const int64_t fsize= url_fsize(s->pb); url_fseek(s->pb, fsize-4, SEEK_SET); size= get_be32(s->pb); url_fseek(s->pb, fsize-3-size, SEEK_SET); if(size == get_be24(s->pb) + 11){ s->duration= get_be24(s->pb) * (int64_t)AV_TIME_BASE / 1000; } url_fseek(s->pb, pos, SEEK_SET); } if(is_audio){ if(!st->codec->channels || !st->codec->sample_rate || !st->codec->bits_per_coded_sample) { st->codec->channels = (flags & FLV_AUDIO_CHANNEL_MASK) == FLV_STEREO ? 2 : 1; st->codec->sample_rate = (44100 << ((flags & FLV_AUDIO_SAMPLERATE_MASK) >> FLV_AUDIO_SAMPLERATE_OFFSET) >> 3); st->codec->bits_per_coded_sample = (flags & FLV_AUDIO_SAMPLESIZE_MASK) ? 16 : 8; } if(!st->codec->codec_id){ flv_set_audio_codec(s, st, flags & FLV_AUDIO_CODECID_MASK); } }else{ size -= flv_set_video_codec(s, st, flags & FLV_VIDEO_CODECID_MASK); } if (st->codec->codec_id == CODEC_ID_AAC || st->codec->codec_id == CODEC_ID_H264) { int type = get_byte(s->pb); size--; if (st->codec->codec_id == CODEC_ID_H264) { int32_t cts = (get_be24(s->pb)+0xff800000)^0xff800000; // sign extension pts = dts + cts; if (cts < 0) { // dts are wrong flv->wrong_dts = 1; av_log(s, AV_LOG_WARNING, "negative cts, previous timestamps might be wrong\n"); } if (flv->wrong_dts) dts = AV_NOPTS_VALUE; } if (type == 0) { if ((ret = flv_get_extradata(s, st, size)) < 0) return ret; if (st->codec->codec_id == CODEC_ID_AAC) { MPEG4AudioConfig cfg; ff_mpeg4audio_get_config(&cfg, st->codec->extradata, st->codec->extradata_size); if (cfg.chan_config > 7) return -1; st->codec->channels = ff_mpeg4audio_channels[cfg.chan_config]; st->codec->sample_rate = cfg.sample_rate; dprintf(s, "mp4a config channels %d sample rate %d\n", st->codec->channels, st->codec->sample_rate); } return AVERROR(EAGAIN); } } ret= av_get_packet(s->pb, pkt, size); if (ret <= 0) { return AVERROR(EIO); } /* note: we need to modify the packet size here to handle the last packet */ pkt->size = ret; pkt->dts = dts; pkt->pts = pts == AV_NOPTS_VALUE ? dts : pts; pkt->stream_index = st->index; if (is_audio || ((flags & FLV_VIDEO_FRAMETYPE_MASK) == FLV_FRAME_KEY)) pkt->flags |= PKT_FLAG_KEY; return ret; } AVInputFormat flv_demuxer = { "flv", NULL_IF_CONFIG_SMALL("FLV format"), sizeof(FLVContext), flv_probe, flv_read_header, flv_read_packet, .extensions = "flv", .value = CODEC_ID_FLV1, };