ffmpeg学习七:avformat_find_stream_info函数源码分析
2016-12-14 17:33
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前面两篇文章分析avformat_open_input和avcodec_open2两个函数,我们所做的函数分析工作都是为了能够很好的理解前面一篇博客:ffmpeg学习四:写第一个程序-视频解码中所给的视频解码的程序。avformat_find_stream_info函数也是视频解码程序中必须要有的函数,因此这篇文章主要来分析这个函数。
从注释来看,avformat_find_stream_info主要是读一些包(packets ),然后从中提取初流的信息。有一些文件格式没有头,比如说MPEG格式的,这个时候,这个函数就很有用,因为它可以从读取到的包中获得到流的信息。在MPEG-2重复帧模式的情况下,该函数还计算真实的帧率。
逻辑文件位置不被此函数更改; 读出来的包会被缓存起来供以后处理。
注释很好的解释了这个函数的功能。我们可以想象一下,既然这个函数的功能是更新流的信息,那么可以猜测它的作用就是更新AVStream这个结构体中的字段了。我们先浏览下这个结构体:
int index:该流的标示。
AVCodecContext *codec:该流的编解码器上下文。
AVRational time_base:时基。通过该值可以把PTS,DTS转化为真正的时间。FFMPEG其他结构体中也有这个字段,但是根据我的经验,只有AVStream中的time_base是可用的。PTS*time_base=真正的时间
start_time:流中第一个pts的时间。
nb_frames:这个流中的帧的数目。
AVRational avg_frame_rate;平均帧率。
int64_t duration:该流的时间长度
AVDictionary *metadata:元数据信息
AVPacket attached_pic:附带的图片。比如说一些MP3,AAC音频文件附带的专辑封面。
重要的数据结构框图如下:
下面来看这个函数的代码,源码在libavformat/utils.c文件中。
这个函数非常的长,我们分开来看.
这里定义了一些参数:
probesize就是探测的大小,为了获得流的信息,这个函数会尝试的读一些包出来,然后分析这些数据,probesize限制 了最大允许读出的数据的大小。
orig_nb_streams是这个文件中的流的数量。普通的电影会包含三个流:音频流,视频流和字幕流。
接着往下看。
这里定义了最大分析时长的限制。max_stream_analyze_duration 等于max_analyze_duration 等于30倍的timebase。此外,如果文件格式为flv或者Mpeg,那么他们会有各自的最大分析时长的限制。
继续往下看:
这个函数有很多次遍历流的循环,这里是第一次。第一次循环做了如下事情:
1.获得编码器上下文环境avctx,设置avctx的time_base,code_id,codec_type,orig_codec_id。
2.如果解析器paser为空,那么会初始化解析器。
3.把解析器中的参数对应的拷贝到编解码器上下文环境中。调用的是avcodec_parameters_to_context函数,这个函数如下:
可见就是根据编码器类型做对应的参数的拷贝。
4根据编码器ID查找编码器.
这里调用的是find_probe_decoder函数,该函数也定义在libavformat/utils.c中。可以看一下查找编解码器的过程:
调用find_decoder函数进一步查找:
如果编码器已经存在就根据编码器类型返回对应的编解码器,否则就根据id进行查找。avcodec_find_decoder函数还是在utils.c文件中:
find_encdec函数也是在utils.c文件中:
这个函数的功能就是遍历一个编解码器的链表,其首个结构是first_avcodec。编解码器链表是我们在av_register_all()函数中注册的。这个逐个匹配每一个编解码器的id,找到后返回对应的编解码器。
5.打开编解码器
打开使用的是avcodec_open2函数。这个函数我们在前一篇博客ffmpeg学习六:avcodec_open2函数源码分析一文中已经分析过了。
所以我们可以总结下第一次遍历所有的流所做的事情:初始化time_base,codec_id等参数,初始化解析器paser,然后对于每一个流,根据codec_id找对对应的编解码器,然后打开编解码器。也就是说,第一次遍历流,使得我们对于每一个流,都有一个编解码器可用。
第二次遍历流是在编解码器已经打开之后,对于每一个流,设置了一些参数。last_dts 是最有的解码时间。fps_first_dts和fps_last_dts 用于帧率的计算。
这个死循环做了如下事情:
1.检查用户有没有请求中断。如果有中断请求,就调用中断回调方法,并且函数返回。
2.再一次遍历流,检查是不是还有编解码器需要进一步处理。这里会检查编解码器的各个参数,如果这些参数都就绪,那么就会返回了,也就没有必要再做进一步分析了。如果还有些编解码器的信息不全,那么这里会继续向下执行。
3.读一帧数据进来。使用read_frame_internal函数。这个函数很复杂,以后在分析。
4.把读出来的数据添加到缓冲区。使用的是add_to_pktbuf函数。然后更新读到的总的数据的大小:read_size += pkt->size;
5.再次更新编解码器上下文环境的参数。调用的是avcodec_parameters_to_context函数,这个函数我们已经分析过了。
6.检查dts的连续性。 如果dts中的差异大于序列中平均分组持续时间的1000倍,我们将其视为不连续。
代码为:
7.更新存储的dts的值。代码为:
8.调用解析器的aplit方法。加入我们的解析器是h.264,那么这个解析器会像下面这样:
这个split方法暂时看不懂,以后再来解析。
9.如果这个时候,还是没有找到足够的信息,那么就会尝试解压一些数据出来并做分析。
首先看注释:
如果仍然没有信息,我们尝试打开编解码器并且解压缩帧。 我们需要在大多数情况下尽量避免做这样的事情,因为很需要很多的时间,并使用更多的内存。 对于MPEG-4,我们需要解压QuickTime。
如果设置AV_CODEC_CAP_CHANNEL_CONF,这将强制解码至少一帧编解码器数据,这确保编解码器初始化通道配置,并且不仅信任来自容器的值
尝试解压一帧的数据使用的是try_decode_frame函数。有些参数,如vidoe的pix_fmt是需要调用h264_decode_frame才可以获取其pix_fmt的。
总结下这个死循环做的工作:其中会检测是不是所有流的信息都已经完备,如果完备就返回,如果完备,就尝试的解压一帧的数据。并从中获取pix_fmt等必须通过解压一帧的数据才能获得的参数。
如果到了文件尾部,又会遍历一次流,检测其编解码器参数,如果其参数不完整,就会再次调用avcodec_open2来初始化编解码器的各个参数。
有一些帧可能在缓存区中,需要把它们flush掉。
再次遍历流,逐个调用avcodec_close方法来关闭流。
这个循环用来处理音频流和视频流。
对视频流而言,首先会获得原始数据的图像格式,然后据此找到对应的编解码器的tag。之后会计算平均帧率。
对音频流而言,基于音频流的服务类型初始化disposition。disposition的作用暂时不知。
estimate_timings函数如下:
主要是调用update_stream_timings函数更新流的时间相关的参数,之后的代码块是打印日志。
update_stream_timings函数定义在libavfomat/utils.c中。
时间相关的计算非常复杂,是在看不懂。这里还计算了波特率: bitrate = (double) filesize * 8.0 * AV_TIME_BASE / (double) ic->duration;
这里很好理解。filesize*8是这个文件的所有bit数。一字节=8bit。
然后比特率=bit数/时间。
这里时间为ic->duration/AV_TIME_BASE 。
最后就是更新各个结构的数据了。AVStream中的AVCodecContext结构体,以及AVStream中AVStreamInternal中的AVCodecContext结构体的数据要统一起来。AVStream中的AVCodecParameters和AVStream中AVStreamInternal中的AVCodecContext结构体的数据也要统一起来。以上主要就是跟新这几个结构体中的数据。
avformat_open_input函数会读文件头,对mp4文件而言,它会解析所有的box。但它知识把读到的结果保存在对应的数据结构下。这个时候,AVStream中的很多字段都是空白的。
avformat_find_stream_info则检测这些重要字段,如果是空白的,就设法填充它们。因为我们解析文件头的时候,已经掌握了大量的信息,avformat_find_stream_info就是通过这些信息来填充自己的成员,当重要的成员都填充完毕后,该函数就返回了。这中情况下,该函数效率很高。但对于某些文件,单纯的从文件头中获取信息是不够的,比如vidoe的pix_fmt是需要调用h264_decode_frame才可以获取其pix_fmt的。因此,这个时候这个函数就会读一些数据进来,然后分析这些数据,并尝试解码这些数据,最终从中提取到所需要的信息。在所有的信息都已经获取到以后,avformat_find_stream_info函数会计算start_time,波特率等信息,其中时间相关的计算很复杂,没能看懂,以后再研究吧。计算结束后会更新相关结构体的数据成员。
虽然由于能力限制,没能够彻底搞懂这个函数的方方面面,但是,通过这次分析,更好的理解了这个函数的功能,对这个函数做的事情有了一定的了解。这也是这次分析的主要收获了,至于那些搞不懂的,只能留着以后慢慢研究了。
一、功能简介
先看看avformat_find_stream_info函数的注释:/** * Read packets of a media file to get stream information. This * is useful for file formats with no headers such as MPEG. This * function also computes the real framerate in case of MPEG-2 repeat * frame mode. * The logical file position is not changed by this function; * examined packets may be buffered for later processing. * * @param ic media file handle * @param options If non-NULL, an ic.nb_streams long array of pointers to * dictionaries, where i-th member contains options for * codec corresponding to i-th stream. * On return each dictionary will be filled with options that were not found. * @return >=0 if OK, AVERROR_xxx on error * * @note this function isn't guaranteed to open all the codecs, so * options being non-empty at return is a perfectly normal behavior. * * @todo Let the user decide somehow what information is needed so that * we do not waste time getting stuff the user does not need. */ int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options);
从注释来看,avformat_find_stream_info主要是读一些包(packets ),然后从中提取初流的信息。有一些文件格式没有头,比如说MPEG格式的,这个时候,这个函数就很有用,因为它可以从读取到的包中获得到流的信息。在MPEG-2重复帧模式的情况下,该函数还计算真实的帧率。
逻辑文件位置不被此函数更改; 读出来的包会被缓存起来供以后处理。
注释很好的解释了这个函数的功能。我们可以想象一下,既然这个函数的功能是更新流的信息,那么可以猜测它的作用就是更新AVStream这个结构体中的字段了。我们先浏览下这个结构体:
二、AVStream结构体解析
/** * Stream structure. * New fields can be added to the end with minor version bumps. * Removal, reordering and changes to existing fields require a major * version bump. * sizeof(AVStream) must not be used outside libav*. */ typedef struct AVStream { int index; /**< stream index in AVFormatContext */ /** * Format-specific stream ID. * decoding: set by libavformat * encoding: set by the user, replaced by libavformat if left unset */ int id; #if FF_API_LAVF_AVCTX /** * @deprecated use the codecpar struct instead */ attribute_deprecated AVCodecContext *codec; #endif void *priv_data; #if FF_API_LAVF_FRAC /** * @deprecated this field is unused */ attribute_deprecated struct AVFrac pts; #endif /** * This is the fundamental unit of time (in seconds) in terms * of which frame timestamps are represented. * * decoding: set by libavformat * encoding: May be set by the caller before avformat_write_header() to * provide a hint to the muxer about the desired timebase. In * avformat_write_header(), the muxer will overwrite this field * with the timebase that will actually be used for the timestamps * written into the file (which may or may not be related to the * user-provided one, depending on the format). */ AVRational time_base; /** * Decoding: pts of the first frame of the stream in presentation order, in stream time base. * Only set this if you are absolutely 100% sure that the value you set * it to really is the pts of the first frame. * This may be undefined (AV_NOPTS_VALUE). * @note The ASF header does NOT contain a correct start_time the ASF * demuxer must NOT set this. */ int64_t start_time; /** * Decoding: duration of the stream, in stream time base. * If a source file does not specify a duration, but does specify * a bitrate, this value will be estimated from bitrate and file size. */ int64_t duration; int64_t nb_frames; ///< number of frames in this stream if known or 0 int disposition; /**< AV_DISPOSITION_* bit field */ enum AVDiscard discard; ///< Selects which packets can be discarded at will and do not need to be demuxed. /** * sample aspect ratio (0 if unknown) * - encoding: Set by user. * - decoding: Set by libavformat. */ AVRational sample_aspect_ratio; AVDictionary *metadata; /** * Average framerate * * - demuxing: May be set by libavformat when creating the stream or in * avformat_find_stream_info(). * - muxing: May be set by the caller before avformat_write_header(). */ AVRational avg_frame_rate; /** * For streams with AV_DISPOSITION_ATTACHED_PIC disposition, this packet * will contain the attached picture. * * decoding: set by libavformat, must not be modified by the caller. * encoding: unused */ AVPacket attached_pic; /** * An array of side data that applies to the whole stream (i.e. the * container does not allow it to change between packets). * * There may be no overlap between the side data in this array and side data * in the packets. I.e. a given side data is either exported by the muxer * (demuxing) / set by the caller (muxing) in this array, then it never * appears in the packets, or the side data is exported / sent through * the packets (always in the first packet where the value becomes known or * changes), then it does not appear in this array. * * - demuxing: Set by libavformat when the stream is created. * - muxing: May be set by the caller before avformat_write_header(). * * Freed by libavformat in avformat_free_context(). * * @see av_format_inject_global_side_data() */ AVPacketSideData *side_data; /** * The number of elements in the AVStream.side_data array. */ int nb_side_data; /** * Flags for the user to detect events happening on the stream. Flags must * be cleared by the user once the event has been handled. * A combination of AVSTREAM_EVENT_FLAG_*. */ int event_flags; #define AVSTREAM_EVENT_FLAG_METADATA_UPDATED 0x0001 ///< The call resulted in updated metadata. /***************************************************************** * All fields below this line are not part of the public API. They * may not be used outside of libavformat and can be changed and * removed at will. * New public fields should be added right above. ***************************************************************** */ /** * Stream information used internally by av_find_stream_info() */ #define MAX_STD_TIMEBASES (30*12+30+3+6) struct { int64_t last_dts; int64_t duration_gcd; int duration_count; int64_t rfps_duration_sum; double (*duration_error)[2][MAX_STD_TIMEBASES]; int64_t codec_info_duration; int64_t codec_info_duration_fields; /** * 0 -> decoder has not been searched for yet. * >0 -> decoder found * <0 -> decoder with codec_id == -found_decoder has not been found */ int found_decoder; int64_t last_duration; /** * Those are used for average framerate estimation. */ int64_t fps_first_dts; int fps_first_dts_idx; int64_t fps_last_dts; int fps_last_dts_idx; } *info; int pts_wrap_bits; /**< number of bits in pts (used for wrapping control) */ // Timestamp generation support: /** * Timestamp corresponding to the last dts sync point. * * Initialized when AVCodecParserContext.dts_sync_point >= 0 and * a DTS is received from the underlying container. Otherwise set to * AV_NOPTS_VALUE by default. */ int64_t first_dts; int64_t cur_dts; int64_t last_IP_pts; int last_IP_duration; /** * Number of packets to buffer for codec probing */ int probe_packets; /** * Number of frames that have been demuxed during av_find_stream_info() */ int codec_info_nb_frames; /* av_read_frame() support */ enum AVStreamParseType need_parsing; struct AVCodecParserContext *parser; /** * last packet in packet_buffer for this stream when muxing. */ struct AVPacketList *last_in_packet_buffer; AVProbeData probe_data; #define MAX_REORDER_DELAY 16 int64_t pts_buffer[MAX_REORDER_DELAY+1]; AVIndexEntry *index_entries; /**< Only used if the format does not support seeking natively. */ int nb_index_entries; unsigned int index_entries_allocated_size; /** * Real base framerate of the stream. * This is the lowest framerate with which all timestamps can be * represented accurately (it is the least common multiple of all * framerates in the stream). Note, this value is just a guess! * For example, if the time base is 1/90000 and all frames have either * approximately 3600 or 1800 timer ticks, then r_frame_rate will be 50/1. * * Code outside avformat should access this field using: * av_stream_get/set_r_frame_rate(stream) */ AVRational r_frame_rate; /** * Stream Identifier * This is the MPEG-TS stream identifier +1 * 0 means unknown */ int stream_identifier; int64_t interleaver_chunk_size; int64_t interleaver_chunk_duration; /** * stream probing state * -1 -> probing finished * 0 -> no probing requested * rest -> perform probing with request_probe being the minimum score to accept. * NOT PART OF PUBLIC API */ int request_probe; /** * Indicates that everything up to the next keyframe * should be discarded. */ int skip_to_keyframe; /** * Number of samples to skip at the start of the frame decoded from the next packet. */ int skip_samples; /** * If not 0, the number of samples that should be skipped from the start of * the stream (the samples are removed from packets with pts==0, which also * assumes negative timestamps do not happen). * Intended for use with formats such as mp3 with ad-hoc gapless audio * support. */ int64_t start_skip_samples; /** * If not 0, the first audio sample that should be discarded from the stream. * This is broken by design (needs global sample count), but can't be * avoided for broken by design formats such as mp3 with ad-hoc gapless * audio support. */ int64_t first_discard_sample; /** * The sample after last sample that is intended to be discarded after * first_discard_sample. Works on frame boundaries only. Used to prevent * early EOF if the gapless info is broken (considered concatenated mp3s). */ int64_t last_discard_sample; /** * Number of internally decoded frames, used internally in libavformat, do not access * its lifetime differs from info which is why it is not in that structure. */ int nb_decoded_frames; /** * Timestamp offset added to timestamps before muxing * NOT PART OF PUBLIC API */ int64_t mux_ts_offset; /** * Internal data to check for wrapping of the time stamp */ int64_t pts_wrap_reference; /** * Options for behavior, when a wrap is detected. * * Defined by AV_PTS_WRAP_ values. * * If correction is enabled, there are two possibilities: * If the first time stamp is near the wrap point, the wrap offset * will be subtracted, which will create negative time stamps. * Otherwise the offset will be added. */ int pts_wrap_behavior; /** * Internal data to prevent doing update_initial_durations() twice */ int update_initial_durations_done; /** * Internal data to generate dts from pts */ int64_t pts_reorder_error[MAX_REORDER_DELAY+1]; uint8_t pts_reorder_error_count[MAX_REORDER_DELAY+1]; /** * Internal data to analyze DTS and detect faulty mpeg streams */ int64_t last_dts_for_order_check; uint8_t dts_ordered; uint8_t dts_misordered; /** * Internal data to inject global side data */ int inject_global_side_data; /** * String containing paris of key and values describing recommended encoder configuration. * Paris are separated by ','. * Keys are separated from values by '='. */ char *recommended_encoder_configuration; /** * display aspect ratio (0 if unknown) * - encoding: unused * - decoding: Set by libavformat to calculate sample_aspect_ratio internally */ AVRational display_aspect_ratio; struct FFFrac *priv_pts; /** * An opaque field for libavformat internal usage. * Must not be accessed in any way by callers. */ AVStreamInternal *internal; /* * Codec parameters associated with this stream. Allocated and freed by * libavformat in avformat_new_stream() and avformat_free_context() * respectively. * * - demuxing: filled by libavformat on stream creation or in * avformat_find_stream_info() * - muxing: filled by the caller before avformat_write_header() */ AVCodecParameters *codecpar; } AVStream;
int index:该流的标示。
AVCodecContext *codec:该流的编解码器上下文。
AVRational time_base:时基。通过该值可以把PTS,DTS转化为真正的时间。FFMPEG其他结构体中也有这个字段,但是根据我的经验,只有AVStream中的time_base是可用的。PTS*time_base=真正的时间
start_time:流中第一个pts的时间。
nb_frames:这个流中的帧的数目。
AVRational avg_frame_rate;平均帧率。
int64_t duration:该流的时间长度
AVDictionary *metadata:元数据信息
AVPacket attached_pic:附带的图片。比如说一些MP3,AAC音频文件附带的专辑封面。
重要的数据结构框图如下:
下面来看这个函数的代码,源码在libavformat/utils.c文件中。
这个函数非常的长,我们分开来看.
三、函数源码分析
part 1 参数定义
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options) { int i, count = 0, ret = 0, j; int64_t read_size; AVStream *st; AVCodecContext *avctx; AVPacket pkt1, *pkt; int64_t old_offset = avio_tell(ic->pb); // new streams might appear, no options for those int orig_nb_streams = ic->nb_streams; int flush_codecs; int64_t max_analyze_duration = ic->max_analyze_duration; int64_t max_stream_analyze_duration; int64_t max_subtitle_analyze_duration; int64_t probesize = ic->probesize; int eof_reached = 0; flush_codecs = probesize > 0; ...
这里定义了一些参数:
probesize就是探测的大小,为了获得流的信息,这个函数会尝试的读一些包出来,然后分析这些数据,probesize限制 了最大允许读出的数据的大小。
orig_nb_streams是这个文件中的流的数量。普通的电影会包含三个流:音频流,视频流和字幕流。
接着往下看。
part 2 设置max_stream_analyze_duration 等
max_stream_analyze_duration = max_analyze_duration; max_subtitle_analyze_duration = max_analyze_duration; if (!max_analyze_duration) { max_stream_analyze_duration = max_analyze_duration = 5*AV_TIME_BASE; max_subtitle_analyze_duration = 30*AV_TIME_BASE; if (!strcmp(ic->iformat->name, "flv")) max_stream_analyze_duration = 90*AV_TIME_BASE; if (!strcmp(ic->iformat->name, "mpeg") || !strcmp(ic->iformat->name, "mpegts")) max_stream_analyze_duration = 7*AV_TIME_BASE; }
这里定义了最大分析时长的限制。max_stream_analyze_duration 等于max_analyze_duration 等于30倍的timebase。此外,如果文件格式为flv或者Mpeg,那么他们会有各自的最大分析时长的限制。
继续往下看:
part 3 第一次遍历流
for (i = 0; i < ic->nb_streams; i++) { const AVCodec *codec; AVDictionary *thread_opt = NULL; st = ic->streams[i]; avctx = st->internal->avctx; if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO || st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE) { /* if (!st->time_base.num) st->time_base = */ if (!avctx->time_base.num) avctx->time_base = st->time_base; } /* check if the caller has overridden the codec id */ #if FF_API_LAVF_AVCTX FF_DISABLE_DEPRECATION_WARNINGS if (st->codec->codec_id != st->internal->orig_codec_id) { st->codecpar->codec_id = st->codec->codec_id; st->codecpar->codec_type = st->codec->codec_type; st->internal->orig_codec_id = st->codec->codec_id; } FF_ENABLE_DEPRECATION_WARNINGS #endif // only for the split stuff if (!st->parser && !(ic->flags & AVFMT_FLAG_NOPARSE) && st->request_probe <= 0) { st->parser = av_parser_init(st->codecpar->codec_id); if (st->parser) { if (st->need_parsing == AVSTREAM_PARSE_HEADERS) { st->parser->flags |= PARSER_FLAG_COMPLETE_FRAMES; } else if (st->need_parsing == AVSTREAM_PARSE_FULL_RAW) { st->parser->flags |= PARSER_FLAG_USE_CODEC_TS; } } else if (st->need_parsing) { av_log(ic, AV_LOG_VERBOSE, "parser not found for codec " "%s, packets or times may be invalid.\n", avcodec_get_name(st->codecpar->codec_id)); } } if (st->codecpar->codec_id != st->internal->orig_codec_id) st->internal->orig_codec_id = st->codecpar->codec_id; ret = avcodec_parameters_to_context(avctx, st->codecpar); if (ret < 0) goto find_stream_info_err; if (st->request_probe <= 0) st->internal->avctx_inited = 1; codec = find_probe_decoder(ic, st, st->codecpar->codec_id); /* Force thread count to 1 since the H.264 decoder will not extract * SPS and PPS to extradata during multi-threaded decoding. */ av_dict_set(options ? &options[i] : &thread_opt, "threads", "1", 0); if (ic->codec_whitelist) av_dict_set(options ? &options[i] : &thread_opt, "codec_whitelist", ic->codec_whitelist, 0); /* Ensure that subtitle_header is properly set. */ if (st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE && codec && !avctx->codec) { if (avcodec_open2(avctx, codec, options ? &options[i] : &thread_opt) < 0) av_log(ic, AV_LOG_WARNING, "Failed to open codec in av_find_stream_info\n"); } // Try to just open decoders, in case this is enough to get parameters. if (!has_codec_parameters(st, NULL) && st->request_probe <= 0) { if (codec && !avctx->codec) if (avcodec_open2(avctx, codec, options ? &options[i] : &thread_opt) < 0) av_log(ic, AV_LOG_WARNING, "Failed to open codec in av_find_stream_info\n"); } if (!options) av_dict_free(&thread_opt); }
这个函数有很多次遍历流的循环,这里是第一次。第一次循环做了如下事情:
1.获得编码器上下文环境avctx,设置avctx的time_base,code_id,codec_type,orig_codec_id。
2.如果解析器paser为空,那么会初始化解析器。
3.把解析器中的参数对应的拷贝到编解码器上下文环境中。调用的是avcodec_parameters_to_context函数,这个函数如下:
int avcodec_parameters_to_context(AVCodecContext *codec, const AVCodecParameters *par) { codec->codec_type = par->codec_type; codec->codec_id = par->codec_id; codec->codec_tag = par->codec_tag; codec->bit_rate = par->bit_rate; codec->bits_per_coded_sample = par->bits_per_coded_sample; codec->bits_per_raw_sample = par->bits_per_raw_sample; codec->profile = par->profile; codec->level = par->level; switch (par->codec_type) { case AVMEDIA_TYPE_VIDEO: codec->pix_fmt = par->format; codec->width = par->width; codec->height = par->height; codec->field_order = par->field_order; codec->color_range = par->color_range; codec->color_primaries = par->color_primaries; codec->color_trc = par->color_trc; codec->colorspace = par->color_space; codec->chroma_sample_location = par->chroma_location; codec->sample_aspect_ratio = par->sample_aspect_ratio; codec->has_b_frames = par->video_delay; break; case AVMEDIA_TYPE_AUDIO: codec->sample_fmt = par->format; codec->channel_layout = par->channel_layout; codec->channels = par->channels; codec->sample_rate = par->sample_rate; codec->block_align = par->block_align; codec->frame_size = par->frame_size; codec->delay = codec->initial_padding = par->initial_padding; codec->trailing_padding = par->trailing_padding; codec->seek_preroll = par->seek_preroll; break; case AVMEDIA_TYPE_SUBTITLE: codec->width = par->width; codec->height = par->height; break; }
可见就是根据编码器类型做对应的参数的拷贝。
4根据编码器ID查找编码器.
这里调用的是find_probe_decoder函数,该函数也定义在libavformat/utils.c中。可以看一下查找编解码器的过程:
static const AVCodec *find_probe_decoder(AVFormatContext *s, const AVStream *st, enum AVCodecID codec_id) { const AVCodec *codec; #if CONFIG_H264_DECODER /* Other parts of the code assume this decoder to be used for h264, * so force it if possible. */ if (codec_id == AV_CODEC_ID_H264) return avcodec_find_decoder_by_name("h264"); #endif codec = find_decoder(s, st, codec_id); if (!codec) return NULL; if (codec->capabilities & AV_CODEC_CAP_AVOID_PROBING) { const AVCodec *probe_codec = NULL; while (probe_codec = av_codec_next(probe_codec)) { if (probe_codec->id == codec_id && av_codec_is_decoder(probe_codec) && !(probe_codec->capabilities & (AV_CODEC_CAP_AVOID_PROBING | AV_CODEC_CAP_EXPERIMENTAL))) { return probe_codec; } } } return codec; }
调用find_decoder函数进一步查找:
static const AVCodec *find_decoder(AVFormatContext *s, const AVStream *st, enum AVCodecID codec_id) { #if FF_API_LAVF_AVCTX FF_DISABLE_DEPRECATION_WARNINGS if (st->codec->codec) return st->codec->codec; FF_ENABLE_DEPRECATION_WARNINGS #endif switch (st->codecpar->codec_type) { case AVMEDIA_TYPE_VIDEO: if (s->video_codec) return s->video_codec; break; case AVMEDIA_TYPE_AUDIO: if (s->audio_codec) return s->audio_codec; break; case AVMEDIA_TYPE_SUBTITLE: if (s->subtitle_codec) return s->subtitle_codec; break; } return avcodec_find_decoder(codec_id); }
如果编码器已经存在就根据编码器类型返回对应的编解码器,否则就根据id进行查找。avcodec_find_decoder函数还是在utils.c文件中:
AVCodec *avcodec_find_decoder(enum AVCodecID id) { return find_encdec(id, 0); }
find_encdec函数也是在utils.c文件中:
static AVCodec *find_encdec(enum AVCodecID id, int encoder) { AVCodec *p, *experimental = NULL; p = first_avcodec; id= remap_deprecated_codec_id(id); while (p) { if ((encoder ? av_codec_is_encoder(p) : av_codec_is_decoder(p)) && p->id == id) { if (p->capabilities & AV_CODEC_CAP_EXPERIMENTAL && !experimental) { experimental = p; } else return p; } p = p->next; } return experimental; }
这个函数的功能就是遍历一个编解码器的链表,其首个结构是first_avcodec。编解码器链表是我们在av_register_all()函数中注册的。这个逐个匹配每一个编解码器的id,找到后返回对应的编解码器。
5.打开编解码器
打开使用的是avcodec_open2函数。这个函数我们在前一篇博客ffmpeg学习六:avcodec_open2函数源码分析一文中已经分析过了。
所以我们可以总结下第一次遍历所有的流所做的事情:初始化time_base,codec_id等参数,初始化解析器paser,然后对于每一个流,根据codec_id找对对应的编解码器,然后打开编解码器。也就是说,第一次遍历流,使得我们对于每一个流,都有一个编解码器可用。
part 4 第二次遍历流
for (i = 0; i < ic->nb_streams; i++) { #if FF_API_R_FRAME_RATE ic->streams[i]->info->last_dts = AV_NOPTS_VALUE; #endif ic->streams[i]->info->fps_first_dts = AV_NOPTS_VALUE; ic->streams[i]->info->fps_last_dts = AV_NOPTS_VALUE; }
第二次遍历流是在编解码器已经打开之后,对于每一个流,设置了一些参数。last_dts 是最有的解码时间。fps_first_dts和fps_last_dts 用于帧率的计算。
part 5 死循环
接下来的这个死循环代码很长。我们回顾一下之前的工作,我们遍历了两次流,设置好了每个流需要的编解码器和计算帧率的参数,初始化了每个流的解析器paser。在介绍这个函数的功能的时候,我们说这个函数会尝试的读一些数据进来,并解析这些数据,从这些数据中进一步获得详细的流的信息。到目前为止我们并没有读任何数据进来,那么接下来,想必就是读数据进来并解码分析数据流了吧。read_size = 0; for (;;) { int analyzed_all_streams; if (ff_check_interrupt(&ic->interrupt_callback)) { ret = AVERROR_EXIT; av_log(ic, AV_LOG_DEBUG, "interrupted\n"); break; } /* check if one codec still needs to be handled */ for (i = 0; i < ic->nb_streams; i++) { int fps_analyze_framecount = 20; st = ic->streams[i]; if (!has_codec_parameters(st, NULL)) break; /* If the timebase is coarse (like the usual millisecond precision * of mkv), we need to analyze more frames to reliably arrive at * the correct fps. */ if (av_q2d(st->time_base) > 0.0005) fps_analyze_framecount *= 2; if (!tb_unreliable(st->internal->avctx)) fps_analyze_framecount = 0; if (ic->fps_probe_size >= 0) fps_analyze_framecount = ic->fps_probe_size; if (st->disposition & AV_DISPOSITION_ATTACHED_PIC) fps_analyze_framecount = 0; /* variable fps and no guess at the real fps */ if (!(st->r_frame_rate.num && st->avg_frame_rate.num) && st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) { int count = (ic->iformat->flags & AVFMT_NOTIMESTAMPS) ? st->info->codec_info_duration_fields/2 : st->info->duration_count; if (count < fps_analyze_framecount) break; } if (st->parser && st->parser->parser->split && !st->internal->avctx->extradata) break; if (st->first_dts == AV_NOPTS_VALUE && !(ic->iformat->flags & AVFMT_NOTIMESTAMPS) && st->codec_info_nb_frames < ((st->disposition & AV_DISPOSITION_ATTACHED_PIC) ? 1 : ic->max_ts_probe) && (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO || st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)) break; } analyzed_all_streams = 0; if (i == ic->nb_streams) { analyzed_all_streams = 1; /* NOTE: If the format has no header, then we need to read some * packets to get most of the streams, so we cannot stop here. */ if (!(ic->ctx_flags & AVFMTCTX_NOHEADER)) { /* If we found the info for all the codecs, we can stop. */ ret = count; av_log(ic, AV_LOG_DEBUG, "All info found\n"); flush_codecs = 0; break; } } /* We did not get all the codec info, but we read too much data. */ if (read_size >= probesize) { ret = count; av_log(ic, AV_LOG_DEBUG, "Probe buffer size limit of %"PRId64" bytes reached\n", probesize); for (i = 0; i < ic->nb_streams; i++) if (!ic->streams[i]->r_frame_rate.num && ic->streams[i]->info->duration_count <= 1 && ic->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO && strcmp(ic->iformat->name, "image2")) av_log(ic, AV_LOG_WARNING, "Stream #%d: not enough frames to estimate rate; " "consider increasing probesize\n", i); break; } /* NOTE: A new stream can be added there if no header in file * (AVFMTCTX_NOHEADER). */ ret = read_frame_internal(ic, &pkt1); if (ret == AVERROR(EAGAIN)) continue; if (ret < 0) { /* EOF or error*/ eof_reached = 1; break; } pkt = &pkt1; if (!(ic->flags & AVFMT_FLAG_NOBUFFER)) { ret = add_to_pktbuf(&ic->internal->packet_buffer, pkt, &ic->internal->packet_buffer_end, 0); if (ret < 0) goto find_stream_info_err; } st = ic->streams[pkt->stream_index]; if (!(st->disposition & AV_DISPOSITION_ATTACHED_PIC)) read_size += pkt->size; avctx = st->internal->avctx; if (!st->internal->avctx_inited) { ret = avcodec_parameters_to_context(avctx, st->codecpar); if (ret < 0) goto find_stream_info_err; st->internal->avctx_inited = 1; } if (pkt->dts != AV_NOPTS_VALUE && st->codec_info_nb_frames > 1) { /* check for non-increasing dts */ if (st->info->fps_last_dts != AV_NOPTS_VALUE && st->info->fps_last_dts >= pkt->dts) { av_log(ic, AV_LOG_DEBUG, "Non-increasing DTS in stream %d: packet %d with DTS " "%"PRId64", packet %d with DTS %"PRId64"\n", st->index, st->info->fps_last_dts_idx, st->info->fps_last_dts, st->codec_info_nb_frames, pkt->dts); st->info->fps_first_dts = st->info->fps_last_dts = AV_NOPTS_VALUE; } /* Check for a discontinuity in dts. If the difference in dts * is more than 1000 times the average packet duration in the * sequence, we treat it as a discontinuity. */ if (st->info->fps_last_dts != AV_NOPTS_VALUE && st->info->fps_last_dts_idx > st->info->fps_first_dts_idx && (pkt->dts - st->info->fps_last_dts) / 1000 > (st->info->fps_last_dts - st->info->fps_first_dts) / (st->info->fps_last_dts_idx - st->info->fps_first_dts_idx)) { av_log(ic, AV_LOG_WARNING, "DTS discontinuity in stream %d: packet %d with DTS " "%"PRId64", packet %d with DTS %"PRId64"\n", st->index, st->info->fps_last_dts_idx, st->info->fps_last_dts, st->codec_info_nb_frames, pkt->dts); st->info->fps_first_dts = st->info->fps_last_dts = AV_NOPTS_VALUE; } /* update stored dts values */ if (st->info->fps_first_dts == AV_NOPTS_VALUE) { st->info->fps_first_dts = pkt->dts; st->info->fps_first_dts_idx = st->codec_info_nb_frames; } st->info->fps_last_dts = pkt->dts; st->info->fps_last_dts_idx = st->codec_info_nb_frames; } if (st->codec_info_nb_frames>1) { int64_t t = 0; int64_t limit; if (st->time_base.den > 0) t = av_rescale_q(st->info->codec_info_duration, st->time_base, AV_TIME_BASE_Q); if (st->avg_frame_rate.num > 0) t = FFMAX(t, av_rescale_q(st->codec_info_nb_frames, av_inv_q(st->avg_frame_rate), AV_TIME_BASE_Q)); if ( t == 0 && st->codec_info_nb_frames>30 && st->info->fps_first_dts != AV_NOPTS_VALUE && st->info->fps_last_dts != AV_NOPTS_VALUE) t = FFMAX(t, av_rescale_q(st->info->fps_last_dts - st->info->fps_first_dts, st->time_base, AV_TIME_BASE_Q)); if (analyzed_all_streams) limit = max_analyze_duration; else if (avctx->codec_type == AVMEDIA_TYPE_SUBTITLE) limit = max_subtitle_analyze_duration; else limit = max_stream_analyze_duration; if (t >= limit) { av_log(ic, AV_LOG_VERBOSE, "max_analyze_duration %"PRId64" reached at %"PRId64" microseconds st:%d\n", limit, t, pkt->stream_index); if (ic->flags & AVFMT_FLAG_NOBUFFER) av_packet_unref(pkt); break; } if (pkt->duration) { if (avctx->codec_type == AVMEDIA_TYPE_SUBTITLE && pkt->pts != AV_NOPTS_VALUE && pkt->pts >= st->start_time) { st->info->codec_info_duration = FFMIN(pkt->pts - st->start_time, st->info->codec_info_duration + pkt->duration); } else st->info->codec_info_duration += pkt->duration; st->info->codec_info_duration_fields += st->parser && st->need_parsing && avctx->ticks_per_frame ==2 ? st->parser->repeat_pict + 1 : 2; } } #if FF_API_R_FRAME_RATE if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) ff_rfps_add_frame(ic, st, pkt->dts); #endif if (st->parser && st->parser->parser->split && !avctx->extradata) { int i = st->parser->parser->split(avctx, pkt->data, pkt->size); if (i > 0 && i < FF_MAX_EXTRADATA_SIZE) { avctx->extradata_size = i; avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE); if (!avctx->extradata) return AVERROR(ENOMEM); memcpy(avctx->extradata, pkt->data, avctx->extradata_size); } } /* If still no information, we try to open the codec and to * decompress the frame. We try to avoid that in most cases as * it takes longer and uses more memory. For MPEG-4, we need to * decompress for QuickTime. * * If AV_CODEC_CAP_CHANNEL_CONF is set this will force decoding of at * least one frame of codec data, this makes sure the codec initializes * the channel configuration and does not only trust the values from * the container. */ try_decode_frame(ic, st, pkt, (options && i < orig_nb_streams) ? &options[i] : NULL); if (ic->flags & AVFMT_FLAG_NOBUFFER) av_packet_unref(pkt); st->codec_info_nb_frames++; count++; }
这个死循环做了如下事情:
1.检查用户有没有请求中断。如果有中断请求,就调用中断回调方法,并且函数返回。
2.再一次遍历流,检查是不是还有编解码器需要进一步处理。这里会检查编解码器的各个参数,如果这些参数都就绪,那么就会返回了,也就没有必要再做进一步分析了。如果还有些编解码器的信息不全,那么这里会继续向下执行。
3.读一帧数据进来。使用read_frame_internal函数。这个函数很复杂,以后在分析。
4.把读出来的数据添加到缓冲区。使用的是add_to_pktbuf函数。然后更新读到的总的数据的大小:read_size += pkt->size;
5.再次更新编解码器上下文环境的参数。调用的是avcodec_parameters_to_context函数,这个函数我们已经分析过了。
6.检查dts的连续性。 如果dts中的差异大于序列中平均分组持续时间的1000倍,我们将其视为不连续。
代码为:
if (st->info->fps_last_dts != AV_NOPTS_VALUE && st->info->fps_last_dts_idx > st->info->fps_first_dts_idx && (pkt->dts - st->info->fps_last_dts) / 1000 > (st->info->fps_last_dts - st->info->fps_first_dts) / (st->info->fps_last_dts_idx - st->info->fps_first_dts_idx)) { av_log(ic, AV_LOG_WARNING, "DTS discontinuity in stream %d: packet %d with DTS " "%"PRId64", packet %d with DTS %"PRId64"\n", st->index, st->info->fps_last_dts_idx, st->info->fps_last_dts, st->codec_info_nb_frames, pkt->dts); st->info->fps_first_dts = st->info->fps_last_dts = AV_NOPTS_VALUE; }
7.更新存储的dts的值。代码为:
if (st->info->fps_first_dts == AV_NOPTS_VALUE) { st->info->fps_first_dts = pkt->dts; st->info->fps_first_dts_idx = st->codec_info_nb_frames; }
8.调用解析器的aplit方法。加入我们的解析器是h.264,那么这个解析器会像下面这样:
AVCodecParser ff_h264_parser = { .codec_ids = { AV_CODEC_ID_H264 }, .priv_data_size = sizeof(H264ParseContext), .parser_init = init, .parser_parse = h264_parse, .parser_close = h264_close, .split = h264_split, };
这个split方法暂时看不懂,以后再来解析。
9.如果这个时候,还是没有找到足够的信息,那么就会尝试解压一些数据出来并做分析。
首先看注释:
如果仍然没有信息,我们尝试打开编解码器并且解压缩帧。 我们需要在大多数情况下尽量避免做这样的事情,因为很需要很多的时间,并使用更多的内存。 对于MPEG-4,我们需要解压QuickTime。
如果设置AV_CODEC_CAP_CHANNEL_CONF,这将强制解码至少一帧编解码器数据,这确保编解码器初始化通道配置,并且不仅信任来自容器的值
尝试解压一帧的数据使用的是try_decode_frame函数。有些参数,如vidoe的pix_fmt是需要调用h264_decode_frame才可以获取其pix_fmt的。
总结下这个死循环做的工作:其中会检测是不是所有流的信息都已经完备,如果完备就返回,如果完备,就尝试的解压一帧的数据。并从中获取pix_fmt等必须通过解压一帧的数据才能获得的参数。
part 6 检查是否到达文件尾部
if (eof_reached) { int stream_index; for (stream_index = 0; stream_index < ic->nb_streams; stream_index++) { st = ic->streams[stream_index]; avctx = st->internal->avctx; if (!has_codec_parameters(st, NULL)) { const AVCodec *codec = find_probe_decoder(ic, st, st->codecpar->codec_id); if (codec && !avctx->codec) { if (avcodec_open2(avctx, codec, (options && stream_index < orig_nb_streams) ? &options[stream_index] : NULL) < 0) av_log(ic, AV_LOG_WARNING, "Failed to open codec in av_find_stream_info\n"); } } // EOF already reached while reading the stream above. // So continue with reoordering DTS with whatever delay we have. if (ic->internal->packet_buffer && !has_decode_delay_been_guessed(st)) { update_dts_from_pts(ic, stream_index, ic->internal->packet_buffer); } } }
如果到了文件尾部,又会遍历一次流,检测其编解码器参数,如果其参数不完整,就会再次调用avcodec_open2来初始化编解码器的各个参数。
part 7 刷新解码器
if (flush_codecs) { AVPacket empty_pkt = { 0 }; int err = 0; av_init_packet(&empty_pkt); for (i = 0; i < ic->nb_streams; i++) { st = ic->streams[i]; /* flush the decoders */ if (st->info->found_decoder == 1) { do { err = try_decode_frame(ic, st, &empty_pkt, (options && i < orig_nb_streams) ? &options[i] : NULL); } while (err > 0 && !has_codec_parameters(st, NULL)); if (err < 0) { av_log(ic, AV_LOG_INFO, "decoding for stream %d failed\n", st->index); } } } }
有一些帧可能在缓存区中,需要把它们flush掉。
part 8 关闭可能可打开的编解码器
for (i = 0; i < ic->nb_streams; i++) { st = ic->streams[i]; avcodec_close(st->internal->avctx); }
再次遍历流,逐个调用avcodec_close方法来关闭流。
part 9 计算rfps
ff_rfps_calculate(ic);part10 再次遍历流
for (i = 0; i < ic->nb_streams; i++) { st = ic->streams[i]; avctx = st->internal->avctx; if (avctx->codec_type == AVMEDIA_TYPE_VIDEO) { if (avctx->codec_id == AV_CODEC_ID_RAWVIDEO && !avctx->codec_tag && !avctx->bits_per_coded_sample) { uint32_t tag= avcodec_pix_fmt_to_codec_tag(avctx->pix_fmt); if (avpriv_find_pix_fmt(avpriv_get_raw_pix_fmt_tags(), tag) == avctx->pix_fmt) avctx->codec_tag= tag; } /* estimate average framerate if not set by demuxer */ if (st->info->codec_info_duration_fields && !st->avg_frame_rate.num && st->info->codec_info_duration) { int best_fps = 0; double best_error = 0.01; if (st->info->codec_info_duration >= INT64_MAX / st->time_base.num / 2|| st->info->codec_info_duration_fields >= INT64_MAX / st->time_base.den || st->info->codec_info_duration < 0) continue; av_reduce(&st->avg_frame_rate.num, &st->avg_frame_rate.den, st->info->codec_info_duration_fields * (int64_t) st->time_base.den, st->info->codec_info_duration * 2 * (int64_t) st->time_base.num, 60000); /* Round guessed framerate to a "standard" framerate if it's * within 1% of the original estimate. */ for (j = 0; j < MAX_STD_TIMEBASES; j++) { AVRational std_fps = { get_std_framerate(j), 12 * 1001 }; double error = fabs(av_q2d(st->avg_frame_rate) / av_q2d(std_fps) - 1); if (error < best_error) { best_error = error; best_fps = std_fps.num; } } if (best_fps) av_reduce(&st->avg_frame_rate.num, &st->avg_frame_rate.den, best_fps, 12 * 1001, INT_MAX); } if (!st->r_frame_rate.num) { if ( avctx->time_base.den * (int64_t) st->time_base.num <= avctx->time_base.num * avctx->ticks_per_frame * (int64_t) st->time_base.den) { st->r_frame_rate.num = avctx->time_base.den; st->r_frame_rate.den = avctx->time_base.num * avctx->ticks_per_frame; } else { st->r_frame_rate.num = st->time_base.den; st->r_frame_rate.den = st->time_base.num; } } if (st->display_aspect_ratio.num && st->display_aspect_ratio.den) { AVRational hw_ratio = { avctx->height, avctx->width }; st->sample_aspect_ratio = av_mul_q(st->display_aspect_ratio, hw_ratio); } } else if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) { if (!avctx->bits_per_coded_sample) avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec_id); // set stream disposition based on audio service type switch (avctx->audio_service_type) { case AV_AUDIO_SERVICE_TYPE_EFFECTS: st->disposition = AV_DISPOSITION_CLEAN_EFFECTS; break; case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED: st->disposition = AV_DISPOSITION_VISUAL_IMPAIRED; break; case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED: st->disposition = AV_DISPOSITION_HEARING_IMPAIRED; break; case AV_AUDIO_SERVICE_TYPE_COMMENTARY: st->disposition = AV_DISPOSITION_COMMENT; break; case AV_AUDIO_SERVICE_TYPE_KARAOKE: st->disposition = AV_DISPOSITION_KARAOKE; break; } } }
这个循环用来处理音频流和视频流。
对视频流而言,首先会获得原始数据的图像格式,然后据此找到对应的编解码器的tag。之后会计算平均帧率。
对音频流而言,基于音频流的服务类型初始化disposition。disposition的作用暂时不知。
part 11 计算时间相关的参数
if (probesize) estimate_timings(ic, old_offset);
estimate_timings函数如下:
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); } if ((!strcmp(ic->iformat->name, "mpeg") || !strcmp(ic->iformat->name, "mpegts")) && file_size && ic->pb->seekable) { /* 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)) { /* 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 { /* 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); } }
主要是调用update_stream_timings函数更新流的时间相关的参数,之后的代码块是打印日志。
update_stream_timings函数定义在libavfomat/utils.c中。
/** * Estimate the stream timings from the one of each components. * * Also computes the global bitrate if possible. */ static void update_stream_timings(AVFormatContext *ic) { int64_t start_time, start_time1, start_time_text, end_time, end_time1, end_time_text; int64_t duration, duration1, filesize; int i; AVStream *st; AVProgram *p; start_time = INT64_MAX; start_time_text = INT64_MAX; end_time = INT64_MIN; end_time_text = INT64_MIN; duration = INT64_MIN; for (i = 0; i < ic->nb_streams; i++) { st = ic->streams[i]; if (st->start_time != AV_NOPTS_VALUE && st->time_base.den) { start_time1 = av_rescale_q(st->start_time, st->time_base, AV_TIME_BASE_Q); if (st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE || st->codecpar->codec_type == AVMEDIA_TYPE_DATA) { if (start_time1 < start_time_text) start_time_text = start_time1; } else start_time = FFMIN(start_time, start_time1); end_time1 = av_rescale_q_rnd(st->duration, st->time_base, AV_TIME_BASE_Q, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); if (end_time1 != AV_NOPTS_VALUE && (end_time1 > 0 ? start_time1 <= INT64_MAX - end_time1 : start_time1 >= INT64_MIN - end_time1)) { end_time1 += start_time1; if (st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE || st->codecpar->codec_type == AVMEDIA_TYPE_DATA) end_time_text = FFMAX(end_time_text, end_time1); else end_time = FFMAX(end_time, end_time1); } for (p = NULL; (p = av_find_program_from_stream(ic, p, i)); ) { if (p->start_time == AV_NOPTS_VALUE || p->start_time > start_time1) p->start_time = start_time1; if (p->end_time < end_time1) p->end_time = end_time1; } } if (st->duration != AV_NOPTS_VALUE) { duration1 = av_rescale_q(st->duration, st->time_base, AV_TIME_BASE_Q); duration = FFMAX(duration, duration1); } } if (start_time == INT64_MAX || (start_time > start_time_text && start_time - start_time_text < AV_TIME_BASE)) start_time = start_time_text; else if (start_time > start_time_text) av_log(ic, AV_LOG_VERBOSE, "Ignoring outlier non primary stream starttime %f\n", start_time_text / (float)AV_TIME_BASE); if (end_time == INT64_MIN || (end_time < end_time_text && end_time_text - end_time < AV_TIME_BASE)) { end_time = end_time_text; } else if (end_time < end_time_text) { av_log(ic, AV_LOG_VERBOSE, "Ignoring outlier non primary stream endtime %f\n", end_time_text / (float)AV_TIME_BASE); } if (start_time != INT64_MAX) { ic->start_time = start_time; if (end_time != INT64_MIN) { if (ic->nb_programs > 1) { for (i = 0; i < ic->nb_programs; i++) { p = ic->programs[i]; if (p->start_time != AV_NOPTS_VALUE && p->end_time > p->start_time) duration = FFMAX(duration, p->end_time - p->start_time); } } else duration = FFMAX(duration, end_time - start_time); } } if (duration != INT64_MIN && duration > 0 && ic->duration == AV_NOPTS_VALUE) { ic->duration = duration; } if (ic->pb && (filesize = avio_size(ic->pb)) > 0 && ic->duration > 0) { /* compute the bitrate */ double bitrate = (double) filesize * 8.0 * AV_TIME_BASE / (double) ic->duration; if (bitrate >= 0 && bitrate <= INT64_MAX) ic->bit_rate = bitrate; } }
时间相关的计算非常复杂,是在看不懂。这里还计算了波特率: bitrate = (double) filesize * 8.0 * AV_TIME_BASE / (double) ic->duration;
这里很好理解。filesize*8是这个文件的所有bit数。一字节=8bit。
然后比特率=bit数/时间。
这里时间为ic->duration/AV_TIME_BASE 。
part 12 更新数据
/* update the stream parameters from the internal codec contexts */ for (i = 0; i < ic->nb_streams; i++) { st = ic->streams[i]; if (st->internal->avctx_inited) { int orig_w = st->codecpar->width; int orig_h = st->codecpar->height; ret = avcodec_parameters_from_context(st->codecpar, st->internal->avctx); if (ret < 0) goto find_stream_info_err; // The decoder might reduce the video size by the lowres factor. if (av_codec_get_lowres(st->internal->avctx) && orig_w) { st->codecpar->width = orig_w; st->codecpar->height = orig_h; } } #if FF_API_LAVF_AVCTX FF_DISABLE_DEPRECATION_WARNINGS ret = avcodec_parameters_to_context(st->codec, st->codecpar); if (ret < 0) goto find_stream_info_err; // The old API (AVStream.codec) "requires" the resolution to be adjusted // by the lowres factor. if (av_codec_get_lowres(st->internal->avctx) && st->internal->avctx->width) { av_codec_set_lowres(st->codec, av_codec_get_lowres(st->internal->avctx)); st->codec->width = st->internal->avctx->width; st->codec->height = st->internal->avctx->height; } if (st->codec->codec_tag != MKTAG('t','m','c','d')) { st->codec->time_base = st->internal->avctx->time_base; st->codec->ticks_per_frame = st->internal->avctx->ticks_per_frame; } st->codec->framerate = st->avg_frame_rate; if (st->internal->avctx->subtitle_header) { st->codec->subtitle_header = av_malloc(st->internal->avctx->subtitle_header_size); if (!st->codec->subtitle_header) goto find_stream_info_err; st->codec->subtitle_header_size = st->internal->avctx->subtitle_header_size; memcpy(st->codec->subtitle_header, st->internal->avctx->subtitle_header, st->codec->subtitle_header_size); } // Fields unavailable in AVCodecParameters st->codec->coded_width = st->internal->avctx->coded_width; st->codec->coded_height = st->internal->avctx->coded_height; st->codec->properties = st->internal->avctx->properties; FF_ENABLE_DEPRECATION_WARNINGS #endif st->internal->avctx_inited = 0;
最后就是更新各个结构的数据了。AVStream中的AVCodecContext结构体,以及AVStream中AVStreamInternal中的AVCodecContext结构体的数据要统一起来。AVStream中的AVCodecParameters和AVStream中AVStreamInternal中的AVCodecContext结构体的数据也要统一起来。以上主要就是跟新这几个结构体中的数据。
四、总结
最后总结一下吧:avformat_open_input函数会读文件头,对mp4文件而言,它会解析所有的box。但它知识把读到的结果保存在对应的数据结构下。这个时候,AVStream中的很多字段都是空白的。
avformat_find_stream_info则检测这些重要字段,如果是空白的,就设法填充它们。因为我们解析文件头的时候,已经掌握了大量的信息,avformat_find_stream_info就是通过这些信息来填充自己的成员,当重要的成员都填充完毕后,该函数就返回了。这中情况下,该函数效率很高。但对于某些文件,单纯的从文件头中获取信息是不够的,比如vidoe的pix_fmt是需要调用h264_decode_frame才可以获取其pix_fmt的。因此,这个时候这个函数就会读一些数据进来,然后分析这些数据,并尝试解码这些数据,最终从中提取到所需要的信息。在所有的信息都已经获取到以后,avformat_find_stream_info函数会计算start_time,波特率等信息,其中时间相关的计算很复杂,没能看懂,以后再研究吧。计算结束后会更新相关结构体的数据成员。
虽然由于能力限制,没能够彻底搞懂这个函数的方方面面,但是,通过这次分析,更好的理解了这个函数的功能,对这个函数做的事情有了一定的了解。这也是这次分析的主要收获了,至于那些搞不懂的,只能留着以后慢慢研究了。
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