ffmpeg系列-解决ffmpeg获取aac音频文件duration不准

这个问题是这样产生的，一同事反应会随机出现ijk获取到的aac文件的duration不准，发来一看，确实不准，在AE或者系统mediaplayer中得到的都是8.4秒（准确时间是MtkAACExtractor: ADTS: duration = 8448000 us），ijk得到的是9.3秒，在播放的时候，在8秒的时候流就结束了，放到编译的ffmpeg中，一看也是9.3秒。

1.分析问题

下面开始分析这个问题，命令行看下这个文件，ffmpeg中获取到的确实是9.3秒

仔细看下红色箭头所指，这个意思是获取到的duration是根据比特率计算的，可能不准确。这种获取音视频info有问题的我们一般可以从avformat_find_stream_info函数开始分析。

这里直接从log开始看，waring出现出现在utils.c/libavformat下

static void estimate_timings_from_bit_rate(AVFormatContext *ic)
{int64_t filesize, duration;int i, show_warning = 0;AVStream *st;av_log(ic, AV_LOG_WARNING,"hxk-->ic->bit_rate:%lld\n",ic->bit_rate);//这里从log可以看到，bitrate也没获取到，bitrate = 0/* if bit_rate is already set, we believe it */if (ic->bit_rate <= 0) {int64_t bit_rate = 0;for (i = 0; i < ic->nb_streams; i++) {st = ic->streams[i];if (st->codecpar->bit_rate <= 0 && st->internal->avctx->bit_rate > 0)st->codecpar->bit_rate = st->internal->avctx->bit_rate;if (st->codecpar->bit_rate > 0) {if (INT64_MAX - st->codecpar->bit_rate < bit_rate) {bit_rate = 0;break;}bit_rate += st->codecpar->bit_rate;} else if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO && st->codec_info_nb_frames > 1) {// If we have a videostream with packets but without a bitrate// then consider the sum not knownbit_rate = 0;break;}}//这里算出来一个bitrateic->bit_rate = bit_rate;av_log(ic, AV_LOG_WARNING,"hxk-->ic->bit_rate:%lld\n",ic->bit_rate);}//从log中可以看到，这里的duration也是0/* if duration is already set, we believe it */av_log(ic, AV_LOG_WARNING,"hxk-->ic->duration:%lld\n",ic->duration);if (ic->duration == AV_NOPTS_VALUE &&ic->bit_rate != 0) {filesize = ic->pb ? avio_size(ic->pb) : 0;av_log(ic, AV_LOG_WARNING,"hxk-->ic->filesize:%lld\n",filesize);if (filesize > ic->internal->data_offset) {filesize -= ic->internal->data_offset;for (i = 0; i < ic->nb_streams; i++) {st      = ic->streams[i];if (   st->time_base.num <= INT64_MAX / ic->bit_rate&& st->duration == AV_NOPTS_VALUE) {//这里根据文件字节*8 /比特率来计算duration，这里cbr这样计算可以计算，但是如果vbr（码率动态）的话就有问题了duration = av_rescale(8 * filesize, st->time_base.den,ic->bit_rate *(int64_t) st->time_base.num);//获取到的duration就不准确了st->duration = duration;show_warning = 1;}}}}if (show_warning)av_log(ic, AV_LOG_WARNING,"Estimating duration from bitrate, this may be inaccurate\n");
}

调用上面这个函数的地方是utils.c/libavofrmat：

static void estimate_timings(AVFormatContext *ic, int64_t old_offset)
{int64_t file_size;/* get the file size, if possible */if (ic->iformat->flags & AVFMT_NOFILE) {file_size = 0;} else {file_size = avio_size(ic->pb);file_size = FFMAX(0, file_size);}av_log(ic, AV_LOG_WARNING, "hxk->ic->iformat->name:%s\n", ic->iformat->name);av_log(ic, AV_LOG_WARNING, "hxk->file_size:%lld\n", file_size);av_log(ic, AV_LOG_WARNING, "hxk->ic->pb->seekable:%d\n", ic->pb->seekable);if ((!strcmp(ic->iformat->name, "mpeg") ||!strcmp(ic->iformat->name, "mpegts")) &&file_size && (ic->pb->seekable & AVIO_SEEKABLE_NORMAL)) {/* get accurate estimate from the PTSes */estimate_timings_from_pts(ic, old_offset);ic->duration_estimation_method = AVFMT_DURATION_FROM_PTS;} else if (has_duration(ic)) {//如果在demuxer中获取到duration了/* at least one component has timings - we use them for all* the components */fill_all_stream_timings(ic);ic->duration_estimation_method = AVFMT_DURATION_FROM_STREAM;} else {//这个文件没有获取到duration，所以走的是这里/* less precise: use bitrate info */estimate_timings_from_bit_rate(ic);ic->duration_estimation_method = AVFMT_DURATION_FROM_BITRATE;}update_stream_timings(ic);{int i;AVStream av_unused *st;for (i = 0; i < ic->nb_streams; i++) {st = ic->streams[i];av_log(ic, AV_LOG_TRACE, "stream %d: start_time: %0.3f duration: %0.3f\n", i,(double) st->start_time * av_q2d(st->time_base),(double) st->duration   * av_q2d(st->time_base));}av_log(ic, AV_LOG_TRACE,"format: start_time: %0.3f duration: %0.3f bitrate=%"PRId64" kb/s\n",(double) ic->start_time / AV_TIME_BASE,(double) ic->duration   / AV_TIME_BASE,(int64_t)ic->bit_rate / 1000);}
}

调用上面这个方法是在avformat_find_stream_info/utils.c/libavformat函数中。

2.解决问题

原因已经知道了，那么可以如何解决这个问题呢？
aac的duration可以如何获取呢？
我们看下android系统中libstagefright框架中aacextractore的实现

AACExtractor.cpp/libstagefrgiht

AACExtractor::AACExtractor(const sp<DataSource> &source, const sp<AMessage> &_meta): mDataSource(source),mInitCheck(NO_INIT),mFrameDurationUs(0) {sp<AMessage> meta = _meta;if (meta == NULL) {String8 mimeType;float confidence;sp<AMessage> _meta;if (!SniffAAC(mDataSource, &mimeType, &confidence, &meta)) {return;}}int64_t offset;CHECK(meta->findInt64("offset", &offset));uint8_t profile, sf_index, channel, header[2];if (mDataSource->readAt(offset + 2, &header, 2) < 2) {return;}
//获取profileprofile = (header[0] >> 6) & 0x3;
//获取采样索引sf_index = (header[0] >> 2) & 0xf;
//获取采样率uint32_t sr = get_sample_rate(sf_index);if (sr == 0) {return;}
//通道channel = (header[0] & 0x1) << 2 | (header[1] >> 6);mMeta = MakeAACCodecSpecificData(profile, sf_index, channel);off64_t streamSize, numFrames = 0;size_t frameSize = 0;int64_t duration = 0;
//获取文件大小if (mDataSource->getSize(&streamSize) == OK) {while (offset < streamSize) {//获取adts每一帧大小if ((frameSize = getAdtsFrameLength(source, offset, NULL)) == 0) {return;}mOffsetVector.push(offset);offset += frameSize;//偏移加加numFrames ++;//计算帧数目}
//***************重点看下这里，这里在下面分析aac文件格式的时候会讲解细致一点*************// Round up and get the durationmFrameDurationUs = (1024 * 1000000ll + (sr - 1)) / sr;duration = numFrames * mFrameDurationUs;//总帧数x一个AAC音频帧的播放时间mMeta->setInt64(kKeyDuration, duration);}mInitCheck = OK;
}

我们再看下getAdtsFrameLength/AACExtractor.cpp/libstagefrgiht函数，这个函数其实就是根据adts头来计算出每一个framesize的大小的

static size_t getAdtsFrameLength(const sp<DataSource> &source, off64_t offset, size_t* headerSize) {
//CRCconst size_t kAdtsHeaderLengthNoCrc = 7;const size_t kAdtsHeaderLengthWithCrc = 9;size_t frameSize = 0;
//同步字uint8_t syncword[2];if (source->readAt(offset, &syncword, 2) != 2) {return 0;}if ((syncword[0] != 0xff) || ((syncword[1] & 0xf6) != 0xf0)) {return 0;}
//0没有crc，1有crcuint8_t protectionAbsent;if (source->readAt(offset + 1, &protectionAbsent, 1) < 1) {return 0;}protectionAbsent &= 0x1;uint8_t header[3];if (source->readAt(offset + 3, &header, 3) < 3) {return 0;}
//获取framesize的大小frameSize = (header[0] & 0x3) << 11 | header[1] << 3 | header[2] >> 5;// protectionAbsent is 0 if there is CRCsize_t headSize = protectionAbsent ? kAdtsHeaderLengthNoCrc : kAdtsHeaderLengthWithCrc;if (headSize > frameSize) {return 0;}if (headerSize != NULL) {*headerSize = headSize;}return frameSize;
}

上面的实现原理就是根据一个AAC原始帧包含一段时间内1024个采样及相关数据。一个AAC音频帧的播放时间=一个AAC帧对应的采样样本的个数/采样率。所以aac音频文件总时间t=总帧数x一个AAC音频帧的播放时间

下面看一下aac的demuxer，在aacdec.c/libavformat下，发现里面连对aidf头的处理都没有，这个先不管了。

AAC格式：

下面我们先简单看下aac的格式：

详细的AAC格式参考下这篇文章吧，实在懒得写。

AAC文件格式与音频文件时长计算

解决问题

下面我们看下ffmpeg中这个格式的demuxer，这个文件封装格式raw ADTS AAC，下面我们看下aacdec.c/libavformat

修改aacdec.c文件，新加函数

//add by hxk
//获取adts frame的帧长
static int getAdtsFrameLength(AVFormatContext *s,int64_t offset,int* headerSize)
{int64_t filesize, position = avio_tell(s->pb);  filesize = avio_size(s->pb);//av_log(NULL, AV_LOG_WARNING, "hxk->getAdtsFrameLength.filesize:%d\n",filesize);const int kAdtsHeaderLengthNoCrc = 7;const int kAdtsHeaderLengthWithCrc = 9;int frameSize = 0;uint8_t syncword[2];avio_seek(s->pb, offset, SEEK_SET);//读取同步字if(avio_read(s->pb,&syncword, 2)!= 2){return 0;}if ((syncword[0] != 0xff) || ((syncword[1] & 0xf6) != 0xf0)) {return 0;}uint8_t protectionAbsent;avio_seek(s->pb, offset+1, SEEK_SET);//读取protectionAbsentif (avio_read(s->pb, &protectionAbsent, 1) < 1) {return 0;}protectionAbsent &= 0x1;uint8_t header[3];
//读取headeravio_seek(s->pb, offset+3, SEEK_SET);if (avio_read(s->pb, &header, 3) < 3) {return 0;}//获取framesizeframeSize = (header[0] & 0x3) << 11 | header[1] << 3 | header[2] >> 5;// protectionAbsent is 0 if there is CRCint headSize = protectionAbsent ? kAdtsHeaderLengthNoCrc : kAdtsHeaderLengthWithCrc;if (headSize > frameSize) {return 0;}if (headerSize != NULL) {*headerSize = headSize;}return frameSize;
}
//根据采样率下标获取采样率
static uint32_t get_sample_rate(const uint8_t sf_index)
{static const uint32_t sample_rates[] ={96000, 88200, 64000, 48000, 44100, 32000,24000, 22050, 16000, 12000, 11025, 8000};if (sf_index < sizeof(sample_rates) / sizeof(sample_rates[0])) {return sample_rates[sf_index];}return 0;
}//add end

修改adts_aac_read_header函数

static int adts_aac_read_header(AVFormatContext *s)
{av_log(NULL, AV_LOG_WARNING, "hxk->adts_aac_read_header!\n");AVStream *st;uint16_t state;st = avformat_new_stream(s, NULL);if (!st)return AVERROR(ENOMEM);st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;st->codecpar->codec_id   = s->iformat->raw_codec_id;st->need_parsing         = AVSTREAM_PARSE_FULL_RAW;ff_id3v1_read(s);if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {int64_t cur = avio_tell(s->pb);ff_ape_parse_tag(s);avio_seek(s->pb, cur, SEEK_SET);}// skip data until the first ADTS frame is foundstate = avio_r8(s->pb);while (!avio_feof(s->pb) && avio_tell(s->pb) < s->probesize) {state = (state << 8) | avio_r8(s->pb);if ((state >> 4) != 0xFFF)continue;avio_seek(s->pb, -2, SEEK_CUR);break;}if ((state >> 4) != 0xFFF)return AVERROR_INVALIDDATA;// LCM of all possible ADTS sample rates//   avpriv_set_pts_info(st, 64, 1, 28224000);
//add by hxk
#if  1//句柄指回起点avio_seek(s->pb, 0, SEEK_SET);uint8_t profile, sf_index, channel, header[2];//文件指针移动到文件起点前2个字节avio_seek(s->pb, 2, SEEK_SET);if (avio_read(s->pb,&header, 2) < 2) {av_log(NULL, AV_LOG_ERROR, "avio_read header error!\n");return 0;}int64_t offset = 0;//获取profileprofile = (header[0] >> 6) & 0x3;st->codecpar->profile = profile;//av_log(NULL, AV_LOG_WARNING, "hxk->profile:%d!\n",profile);sf_index = (header[0] >> 2) & 0xf;//获取采样率uint32_t sr = get_sample_rate(sf_index);//av_log(NULL, AV_LOG_WARNING, "hxk->samplerate:%d!\n",sr);if (sr == 0) {av_log(NULL, AV_LOG_ERROR, "avio_read read sampletare error!\n");return 0;}//赋值给codec参数st->codecpar->sample_rate = sr;//获取通道channel = (header[0] & 0x1) << 2 | (header[1] >> 6);if (channel == 0) {av_log(NULL, AV_LOG_ERROR, "adts_aac_read_header read channel error!\n");return 0;}//赋值给codec 参数st->codecpar->channels = channel;//av_log(NULL, AV_LOG_WARNING, "hxk->channel:%d!\n",channel);sf_index = (header[0] >> 2) & 0xf;int frameSize = 0;int64_t mFrameDurationUs = 0;int64_t duration = 0;//采样率赋值给codecst->codecpar->sample_rate = sr;int64_t streamSize, numFrames = 0;avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);//获取文件大小streamSize =  avio_size(s->pb);// av_log(NULL, AV_LOG_WARNING, "hxk->streamSize:%d!\n",streamSize);if (streamSize > 0) {while (offset < streamSize) {if ((frameSize = getAdtsFrameLength(s, offset, NULL)) == 0) {return 0;}offset += frameSize;//偏移加加numFrames ++;//帧数加加，获取总帧数//    av_log(NULL, AV_LOG_WARNING, "hxk->frameSize:%d!\n",frameSize);}// av_log(NULL, AV_LOG_WARNING, "hxk->numFrames:%lld!\n",numFrames);// Round up and get the duration，计算每一帧时间mFrameDurationUs = (1024 * 1000000ll + (sr - 1)) / sr;duration = numFrames * mFrameDurationUs;//us//时间基转换avstream的，us单位(AV_TIME_BASE_Q)转avstream的时间基duration = av_rescale_q(duration,AV_TIME_BASE_Q, st->time_base);st->duration = duration;//   av_log(NULL, AV_LOG_WARNING, "hxk->duration:%d!\n",duration);}#endif
//add endreturn 0;
}

这样在demuxer中就获得了duration，在上面的estimate_timings函数中就直接走has_duration这个判断了，得到的duration也是比较准确的了。

运行一下修改后的代码，从下图可以看到时间已经改变了，和android中MtkAACExtractor获取的duration是一样的了。

后续

正满心欢喜解决了问题后，把改动的代码移植到ijk上的时候，发现不能播放，没报任何错误，文件info读取都是正确的，seek一下的时候报了这么一行错误

IJKMEDIA: /storage/emulated/0/3ee807175fc2488d8264ac014ccf55ff.aac: error while seeking

原来忘记把句柄置回去了
修改如下：

static int adts_aac_read_header(AVFormatContext *s)
{AVStream *st;uint16_t state;st = avformat_new_stream(s, NULL);if (!st)return AVERROR(ENOMEM);st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;st->codecpar->codec_id   = s->iformat->raw_codec_id;st->need_parsing         = AVSTREAM_PARSE_FULL_RAW;ff_id3v1_read(s);if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {int64_t cur = avio_tell(s->pb);ff_ape_parse_tag(s);avio_seek(s->pb, cur, SEEK_SET);}// skip data until the first ADTS frame is foundstate = avio_r8(s->pb);while (!avio_feof(s->pb) && avio_tell(s->pb) < s->probesize) {state = (state << 8) | avio_r8(s->pb);if ((state >> 4) != 0xFFF)continue;avio_seek(s->pb, -2, SEEK_CUR);break;}if ((state >> 4) != 0xFFF)return AVERROR_INVALIDDATA;// LCM of all possible ADTS sample rates// avpriv_set_pts_info(st, 64, 1, 28224000);//add by hxk
#if  1avio_seek(s->pb, 0, SEEK_SET);uint8_t profile, sf_index, channel, header[2];avio_seek(s->pb, 2, SEEK_SET);if (avio_read(s->pb,&header, 2) < 2) {av_log(NULL, AV_LOG_ERROR, "avio_read header error!\n");return 0;}int64_t offset = 0;profile = (header[0] >> 6) & 0x3;st->codecpar->profile = profile;sf_index = (header[0] >> 2) & 0xf;uint32_t sr = get_sample_rate(sf_index);if (sr == 0) {av_log(NULL, AV_LOG_ERROR, "adts_aac_read_header read sampletare error!\n");return 0;}st->codecpar->sample_rate = sr;channel = (header[0] & 0x1) << 2 | (header[1] >> 6);if(channel == 0) {av_log(NULL, AV_LOG_ERROR, "adts_aac_read_header read channel error!\n");return 0;}st->codecpar->channels = channel;sf_index = (header[0] >> 2) & 0xf;int frameSize = 0;int64_t mFrameDurationUs = 0;int64_t duration = 0;st->codecpar->sample_rate = sr;int64_t streamSize, numFrames = 0;avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);streamSize =  avio_size(s->pb);//av_log(NULL, AV_LOG_WARNING, "hxk->streamSize:%d!\n",streamSize);if (streamSize > 0) {while (offset < streamSize) {if ((frameSize = getAdtsFrameLength(s, offset, NULL)) == 0) {return 0;}offset += frameSize;numFrames ++;//av_log(NULL, AV_LOG_WARNING, "hxk->frameSize:%d!\n",frameSize);}// av_log(NULL, AV_LOG_WARNING, "hxk->numFrames:%lld!\n",numFrames);// Round up and get the durationmFrameDurationUs = (1024 * 1000000ll + (sr - 1)) / sr;duration = numFrames * mFrameDurationUs;//usduration = av_rescale_q(duration,AV_TIME_BASE_Q, st->time_base);st->duration = duration;//av_log(NULL, AV_LOG_WARNING, "hxk->duration:%d!\n",duration);}//置回句柄avio_seek(s->pb, 0, SEEK_SET);#endif//add endreturn 0;
}

嗯，可以获取正确的时间来正常播放了。