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解决ffmpeg获取AAC音频文件duration不准

时间:2022-09-26 09:37:43

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解决ffmpeg获取AAC音频文件duration不准

最近测试提出了一个bug,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3m48s(准确时间是MMParserExtractor: ADTS: duration = 228010580us,如下图),ijk得到的是2m54s,在播放的时候,在2m54s的时候流就结束了,放到编译的ffmpeg中, Duration:居然是00:03:13.07,但是VLC是3m53s,这个文件也是奇葩了!其他播放器暂时不去讨论,现在只希望做到MMParserExtractor与IJKPlayer获得的时长一直即可!

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3m13s

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, "-->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, "-->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,"-->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,"-->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, "->ic->iformat->name:%s\n", ic->iformat->name);av_log(ic, AV_LOG_WARNING, "->file_size:%lld\n", file_size);av_log(ic, AV_LOG_WARNING, "->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、解决方案探究

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::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格式介绍

首先需要了解的是AAC文件格式有ADIF和ADTS两种,其中ADIF(Audio Data Interchange Format 音频数据交换格式)的特征是解码必须在明确定义的开始处进行,不能从数据流中间开始;而ADTS(Audio Data Transport Stream 音频数据传输流)则相反,这种格式的特征是有同步字,解码可以在这个流中任何位置开始,正如它的名字一样,这是一种和TS流类似的格式。

ADTS格式中每一帧都有头信息,具备流特征,适合于网络传输与处理,而ADIF只有一个统一的头,并且这两种格式的header格式也是不同的。目前主流使用的都是ADTS格式。

ADTS AAC文件格式如下

详细的AAC格式参考下这篇文章吧

AAC文件格式与音频文件时长计算

获取每帧时长:ffmpeg能正确读到每帧的nb_samples和总体的sample_rate,那么两者相除就是每帧的时长了。

AAC:帧大小1024个sample,采样率为44100Hz ,帧播放时长:acc dur=1024/44100 = 0.02322s=23.22ms

那么如何才能获取准确的时长呢?应该是通过adts frame header取总帧数*每帧时长的值作为duration。

3、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

//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){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 M#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;sf_index = (header[0] >> 2) & 0xf;//获取采样率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;}//赋值给codec 参数st->codecpar->channels = 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);if (streamSize > 0) {while (offset < streamSize) {if ((frameSize = getAdtsFrameLength(s, offset, NULL)) == 0) {goto end;}offset += frameSize;//帧数加加,获取总帧数numFrames ++;}end:av_log(NULL, AV_LOG_WARNING, "---streamSize:%lld,numFrames:%lld!---\n",streamSize, numFrames);// Round up and get the duration,计算每一帧时间mFrameDurationUs = (1024 * 1000000ll + (sr - 1)) / sr;av_log(NULL, AV_LOG_WARNING, "---mFrameDurationUs:%lld!---\n",mFrameDurationUs);duration = numFrames * mFrameDurationUs; //usduration = av_rescale_q(duration,AV_TIME_BASE_Q, st->time_base);st->duration = duration;av_log(NULL, AV_LOG_WARNING, "-------duration:%d------!\n",duration);}//置回句柄avio_seek(s->pb, 0, SEEK_SET);#endif//add endreturn 0;}

本来参照一朵桃花压海棠的博客,这里是return 0的,经测试,部分aac文件无法播放,后来改成上文中的goto了

if ((frameSize = getAdtsFrameLength(s, offset, NULL)) == 0) {return 0;}

目前测试没有问题,能够正常seek与播放!

参考链接:ffmpeg系列-解决ffmpeg获取aac音频文件duration不准_一朵桃花压海棠的博客-CSDN博客_ffmpeg 音频duration

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