ffmpeg源码分析_FFmpeg的FFplay框架分析

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最新推荐文章于 2024-09-21 21:34:03 发布

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文章标签： ffmpeg源码分析

本文链接：https://blog.csdn.net/weixin_30686891/article/details/113071706

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本文深入分析FFmpeg自带播放器ffplay的框架结构，包括数据读取、AVpacket缓存队列、解码、输出、音视频同步等模块。文章详细阐述了各模块的功能，如数据读取线程的工作流程、packet队列和frame队列的设计，以及音视频解码和播放的过程。此外，还讨论了音视频同步、音频处理和播放器控制等方面的内容。

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1.框架分析

ffplay.c是FFmpeg源码⾃带的播放器，调⽤FFmpeg和SDL API实现⼀个⾮常有⽤的播放器。例如哔哩哔哩著名开源项⽬ijkplayer也是基于ffplay.c进⾏⼆次开发。ffplay实现了播放器的主体功能，掌握其原理对于我们独⽴开发播放器⾮常有帮助。看看整体的框架，如下图：

从整体上，分这几个大的模块，数据读取，AVpacket缓存队列，音频解码，视频解码，AVframe缓存队列，声音和视频输出，音视频同步等模块组成。

首先进入stream_open，stream_open主要的工作是创建音视频解码前和后的数据缓存队列，初始化时钟，包括音频，视频，外部时钟等，初始化数据读取线程read_thread。源码如下：

static VideoState *stream_open(const char *filename, AVInputFormat *iformat){    VideoState *is;    is = av_mallocz(sizeof(VideoState));    if (!is)        return NULL;    is->filename = av_strdup(filename);    if (!is->filename)        goto fail;    is->iformat = iformat;    is->ytop    = 0;    is->xleft   = 0;    /* start video display */  //初始化帧队列    if (frame_queue_init(&is->pictq, &is->videoq, VIDEO_PICTURE_QUEUE_SIZE, 1) < 0)        goto fail;    if (frame_queue_init(&is->subpq, &is->subtitleq, SUBPICTURE_QUEUE_SIZE, 0) < 0)        goto fail;    if (frame_queue_init(&is->sampq, &is->audioq, SAMPLE_QUEUE_SIZE, 1) < 0)        goto fail;//初始化packet队列    if (packet_queue_init(&is->videoq) < 0 ||        packet_queue_init(&is->audioq) < 0 ||        packet_queue_init(&is->subtitleq) < 0)        goto fail;    if (!(is->continue_read_thread = SDL_CreateCond())) {        av_log(NULL, AV_LOG_FATAL, "SDL_CreateCond(): %s", SDL_GetError());        goto fail;    }//初始化视频时钟    init_clock(&is->vidclk, &is->videoq.serial);  //初始化音频时钟    init_clock(&is->audclk, &is->audioq.serial);  //初始化外部时钟    init_clock(&is->extclk, &is->extclk.serial);    is->audio_clock_serial = -1;    if (startup_volume < 0)        av_log(NULL, AV_LOG_WARNING, "-volume=%d < 0, setting to 0", startup_volume);    if (startup_volume > 100)        av_log(NULL, AV_LOG_WARNING, "-volume=%d > 100, setting to 100", startup_volume);    startup_volume = av_clip(startup_volume, 0, 100);    startup_volume = av_clip(SDL_MIX_MAXVOLUME * startup_volume / 100, 0, SDL_MIX_MAXVOLUME);    is->audio_volume = startup_volume;    is->muted = 0;    is->av_sync_type = av_sync_type;  //初始化数据读取线程    is->read_tid     = SDL_CreateThread(read_thread, "read_thread", is);    if (!is->read_tid) {        av_log(NULL, AV_LOG_FATAL, "SDL_CreateThread(): %s", SDL_GetError());fail:        stream_close(is);        return NULL;    }    return is;}

看看数据读取线程read_thread做了什么？

大致就是一个解封转过程，然后把解封装的packet放到packet队列。分别调用avformat_alloc_context(分配解封装上下文)，avforamt_open_input(打开文件或网络流，内存数据)，avformat_find_stream_info(找到相关流信息)，stream_component_open(打开音视频流)，av_read_frame(解封转)，packet_queue_put(放入到帧队列)。源码如下，详细请看注释：

typedef struct MyAVPacketList {    AVPacket pkt;    struct MyAVPacketList *next;    int serial;} MyAVPacketList;typedef struct PacketQueue {  //链表    MyAVPacketList *first_pkt, *last_pkt;  //多少个packet    int nb_packets;  //每个packet大小    int size;  //每个packet时长    int64_t duration;  //请求标志    int abort_request;    int serial;  //锁    SDL_mutex *mutex;  //条件变量    SDL_cond *cond;} PacketQueue;

然后把解封装的数据，放到packet队列。看看packet的封装结构，实际就是一个链表。详细源码如下：

typedef struct MyAVPacketList {    AVPacket pkt;    struct MyAVPacketList *next;    int serial;} MyAVPacketList;typedef struct PacketQueue {  //链表    MyAVPacketList *first_pkt, *last_pkt;  //多少个packet    int nb_packets;  //每个packet大小    int size;  //每个packet时长    int64_t duration;  //请求标志    int abort_request;    int serial;  //锁    SDL_mutex *mutex;  //条件变量    SDL_cond *cond;} PacketQueue;

视频解码线程，然后就从video的packet 队列取数据，看看做了什么？大致就是使用packet_queue_get去取packet，然后发送解码，avcode_send_packet，avcode_receive_frame，并刷新解码后的frame队列。详细看看源码，如下：

可以看出ffplay.c还支持过滤器功能，功能还是很全。

取数据。

可以看出音视频和subtitle的所调用的函数是不一样。

解码：

解码完后的数据，插入frame队列。注意，视频和音频，都各自有自己frame队列。

涉及到线程安全，就要加锁。

音频基本上也是走的这个流程。就不再叙述了。

最后就是显示模块了，取出数据，然后根据当前能支持的播放格式和尺寸，看看是否要格式转换，也就是是否要调用sws_scale。

static void video_image_display(VideoState *is){    Frame *vp;    Frame *sp = NULL;    SDL_Rect rect;//从frame队列取数据    vp = frame_queue_peek_last(&is->pictq);    if (is->subtitle_st) {        if (frame_queue_nb_remaining(&is->subpq) > 0) {            sp = frame_queue_peek(&is->subpq);            if (vp->pts >= sp->pts + ((float) sp->sub.start_display_time / 1000)) {                if (!sp->uploaded) {                    uint8_t* pixels[4];                    int pitch[4];                    int i;                    if (!sp->width || !sp->height) {                        sp->width = vp->width;                        sp->height = vp->height;                    }                    if (realloc_texture(&is->sub_texture, SDL_PIXELFORMAT_ARGB8888, sp->width, sp->height, SDL_BLENDMODE_BLEND, 1) < 0)                        return;                    for (i = 0; i < sp->sub.num_rects; i++) {                        AVSubtitleRect *sub_rect = sp->sub.rects[i];                        sub_rect->x = av_clip(sub_rect->x, 0, sp->width );                        sub_rect->y = av_clip(sub_rect->y, 0, sp->height);                        sub_rect->w = av_clip(sub_rect->w, 0, sp->width  - sub_rect->x);                        sub_rect->h = av_clip(sub_rect->h, 0, sp->height - sub_rect->y);                        is->sub_convert_ctx = sws_getCachedContext(is->sub_convert_ctx,                            sub_rect->w, sub_rect->h, AV_PIX_FMT_PAL8,                            sub_rect->w, sub_rect->h, AV_PIX_FMT_BGRA,                            0, NULL, NULL, NULL);                        if (!is->sub_convert_ctx) {                            av_log(NULL, AV_LOG_FATAL, "Cannot initialize the conversion context");                            return;                        }                        if (!SDL_LockTexture(is->sub_texture, (SDL_Rect *)sub_rect, (void **)pixels, pitch)) {                         //格式转换                          sws_scale(is->sub_convert_ctx, (const uint8_t * const *)sub_rect->data, sub_rect->linesize,                                      0, sub_rect->h, pixels, pitch);                            SDL_UnlockTexture(is->sub_texture);                        }                    }                    sp->uploaded = 1;                }            } else                sp = NULL;        }    }//计算显示区域    calculate_display_rect(&rect, is->xleft, is->ytop, is->width, is->height, vp->width, vp->height, vp->sar);    if (!vp->uploaded) {        if (upload_texture(&is->vid_texture, vp->frame, &is->img_convert_ctx) < 0)            return;        vp->uploaded = 1;        vp->flip_v = vp->frame->linesize[0] < 0;    }    set_sdl_yuv_conversion_mode(vp->frame);    SDL_RenderCopyEx(renderer, is->vid_texture, NULL, &rect, 0, NULL, vp->flip_v ? SDL_FLIP_VERTICAL : 0);    set_sdl_yuv_conversion_mode(NULL);    if (sp) {#if USE_ONEPASS_SUBTITLE_RENDER        SDL_RenderCopy(renderer, is->sub_texture, NULL, &rect);#else        int i;        double xratio = (double)rect.w / (double)sp->width;        double yratio = (double)rect.h / (double)sp->height;        for (i = 0; i < sp->sub.num_rects; i++) {            SDL_Rect *sub_rect = (SDL_Rect*)sp->sub.rects[i];            SDL_Rect target = {.x = rect.x + sub_rect->x * xratio,                               .y = rect.y + sub_rect->y * yratio,                               .w = sub_rect->w * xratio,                               .h = sub_rect->h * yratio};            SDL_RenderCopy(renderer, is->sub_texture, sub_rect, &target);        }#endif    }}

static int audio_decode_frame(VideoState *is){    int data_size, resampled_data_size;    int64_t dec_channel_layout;    av_unused double audio_clock0;    int wanted_nb_samples;    Frame *af;    if (is->paused)        return -1;    do {#if defined(_WIN32)        while (frame_queue_nb_remaining(&is->sampq) == 0) {            if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)                return -1;            av_usleep (1000);        }#endif        if (!(af = frame_queue_peek_readable(&is->sampq)))            return -1;        frame_queue_next(&is->sampq);    } while (af->serial != is->audioq.serial);    data_size = av_samples_get_buffer_size(NULL, af->frame->channels,                                           af->frame->nb_samples,                                           af->frame->format, 1);    dec_channel_layout =        (af->frame->channel_layout && af->frame->channels == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?        af->frame->channel_layout : av_get_default_channel_layout(af->frame->channels);    wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);    if (af->frame->format        != is->audio_src.fmt            ||        dec_channel_layout       != is->audio_src.channel_layout ||        af->frame->sample_rate   != is->audio_src.freq           ||        (wanted_nb_samples       != af->frame->nb_samples && !is->swr_ctx)) {        swr_free(&is->swr_ctx);        is->swr_ctx = swr_alloc_set_opts(NULL,                                         is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,                                         dec_channel_layout,           af->frame->format, af->frame->sample_rate,                                         0, NULL);        if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {            av_log(NULL, AV_LOG_ERROR,                   "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!",                    af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), af->frame->channels,                    is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);            swr_free(&is->swr_ctx);            return -1;        }        is->audio_src.channel_layout = dec_channel_layout;        is->audio_src.channels       = af->frame->channels;        is->audio_src.freq = af->frame->sample_rate;        is->audio_src.fmt = af->frame->format;    }    if (is->swr_ctx) {        const uint8_t **in = (const uint8_t **)af->frame->extended_data;        uint8_t **out = &is->audio_buf1;        int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;        int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);        int len2;        if (out_size < 0) {            av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed");            return -1;        }        if (wanted_nb_samples != af->frame->nb_samples) {            if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,                                        wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {                av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed");                return -1;            }        }        av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);        if (!is->audio_buf1)            return AVERROR(ENOMEM);        len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);        if (len2 < 0) {            av_log(NULL, AV_LOG_ERROR, "swr_convert() failed");            return -1;        }        if (len2 == out_count) {            av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small");            if (swr_init(is->swr_ctx) < 0)                swr_free(&is->swr_ctx);        }        is->audio_buf = is->audio_buf1;        resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);    } else {        is->audio_buf = af->frame->data[0];        resampled_data_size = data_size;    }    audio_clock0 = is->audio_clock;    /* update the audio clock with the pts */    if (!isnan(af->pts))        is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;    else        is->audio_clock = NAN;    is->audio_clock_serial = af->serial;#ifdef DEBUG    {        static double last_clock;        printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f",               is->audio_clock - last_clock,               is->audio_clock, audio_clock0);        last_clock = is->audio_clock;    }#endif    return resampled_data_size;}

音频也是类似的情况，调用SDL去播放声音。

总体上总结下：

(1)数据读取线程主要的工作:

打开媒体⽂件。

打开对应码流的decoder以及初始化对应的audio、video、subtitle输出。

创建decoder线程，audio、video和subtitle的解码线程独⽴。

调⽤av_read_frame读取packet，并根据steam_index放⼊不同stream对应的packet队列。

(2)⾳频解码

从packet queue读取packet，解出frame后放⼊frame queue。

(3)视频解码

从packet queue读取packet，解出frame后放⼊frame queue。

(4)字幕解码