视频同步播放器(简单的)

/* * SDL_Lesson.c * * Created on: Aug 12, 2014 * Author: clarck */ #define inline _inline #include "sdl/SDL.h" #include "sdl/SDL_thread.h" #include "sdl/SDL_events.h" //#include "../include/logger.h" #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libavutil/pixfmt.h" #include "libswscale/swscale.h" #include "libswresample/swresample.h" #define SDL_AUDIO_BUFFER_SIZE 1024 #define MAX_AUDIO_SIZE (5 * 16 * 1024) #define MAX_VIDEO_SIZE (5 * 256 * 1024) #define FF_ALLOC_EVENT (SDL_USEREVENT) #define FF_REFRESH_EVENT (SDL_USEREVENT + 1) #define FF_QUIT_EVENT (SDL_USEREVENT + 2) #define VIDEO_PICTURE_QUEUE_SIZE 1 #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio #define AV_SYNC_THRESHOLD 0.01 #define AV_NOSYNC_THRESHOLD 10.0 typedef struct PacketQueue { AVPacketList *first_pkt, *last_pkt; int nb_packets; int size; SDL_mutex *mutex; SDL_cond *cond; } PacketQueue; typedef struct VideoPicture { SDL_Window *screen; SDL_Renderer *renderer; SDL_Texture *bmp; AVFrame* rawdata; int width, height; /*source height & width*/ int allocated; double pts; } VideoPicture; typedef struct VideoState { char filename[1024]; AVFormatContext *ic; int videoStream, audioStream; AVStream *audio_st; AVFrame *audio_frame; PacketQueue audioq; unsigned int audio_buf_size; unsigned int audio_buf_index; AVPacket audio_pkt; uint8_t *audio_pkt_data; int audio_pkt_size; int audio_hw_buf_size;// Tian uint8_t *audio_buf; DECLARE_ALIGNED(16,uint8_t,audio_buf2) [AVCODEC_MAX_AUDIO_FRAME_SIZE * 4]; enum AVSampleFormat audio_src_fmt; enum AVSampleFormat audio_tgt_fmt; int audio_src_channels; int audio_tgt_channels; int64_t audio_src_channel_layout; int64_t audio_tgt_channel_layout; int audio_src_freq; int audio_tgt_freq; struct SwrContext *swr_ctx; double audio_clock; double frame_timer; double frame_last_pts; double frame_last_delay; double video_clock; AVStream *video_st; PacketQueue videoq; VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE]; int pictq_size, pictq_rindex, pictq_windex; SDL_mutex *pictq_mutex; SDL_cond *pictq_cond; SDL_Thread *parse_tid; SDL_Thread *audio_tid; SDL_Thread *video_tid; AVIOContext *io_ctx; struct SwsContext *sws_ctx; int quit; } VideoState; VideoState *global_video_state; uint64_t global_video_pkt_pts = AV_NOPTS_VALUE; void packet_queue_init(PacketQueue *q) { memset(q, 0, sizeof(PacketQueue)); q->mutex = SDL_CreateMutex(); q->cond = SDL_CreateCond(); } int packet_queue_put(PacketQueue *q, AVPacket *pkt) { AVPacketList *pkt1; pkt1 = (AVPacketList *) av_malloc(sizeof(AVPacketList)); if (!pkt1) { return -1; } pkt1->pkt = *pkt; pkt1->next = NULL; SDL_LockMutex(q->mutex); if (!q->last_pkt) { q->first_pkt = pkt1; } else { q->last_pkt->next = pkt1; } q->last_pkt = pkt1; q->nb_packets++; q->size += pkt1->pkt.size; SDL_CondSignal(q->cond); SDL_UnlockMutex(q->mutex); return 0; } static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) { AVPacketList *pkt1; int ret; SDL_LockMutex(q->mutex); for (;;) { if (global_video_state->quit) { ret = -1; break; } pkt1 = q->first_pkt; if (pkt1) { q->first_pkt = pkt1->next; if (!q->first_pkt) { q->last_pkt = NULL; } q->nb_packets--; q->size -= pkt1->pkt.size; *pkt = pkt1->pkt; av_free(pkt1); ret = 1; break; } else if (!block) { ret = 0; break; } else { SDL_CondWait(q->cond, q->mutex); } } SDL_UnlockMutex(q->mutex); return ret; } double get_audio_clock(VideoState *is) { double pts; int hw_buf_size, bytes_per_sec, n; pts = is->audio_clock; /* maintained in the audio thread */ hw_buf_size = is->audio_buf_size - is->audio_buf_index; bytes_per_sec = 0; n = is->audio_st->codec->channels * 2; if (is->audio_st) { bytes_per_sec = is->audio_st->codec->sample_rate * n; } if (bytes_per_sec) { pts -= (double)hw_buf_size / bytes_per_sec; } return pts; } int audio_decode_frame(VideoState *is) { int len1=0, len2=0, decoded_data_size; AVPacket *pkt = &is->audio_pkt; // double audio_clock0; int got_frame = 0; int64_t dec_channel_layout; int wanted_nb_samples, resampled_data_size; for (;;) { while (is->audio_pkt_size > 0) { if (!is->audio_frame) { if (!(is->audio_frame = av_frame_alloc())) { return AVERROR(ENOMEM); } } else av_frame_unref(is->audio_frame); // avcodec_get_frame_defaults(is->audio_frame); /** * 当AVPacket中装得是音频时，有可能一个AVPacket中有多个AVFrame， * 而某些解码器只会解出第一个AVFrame，这种情况我们必须循环解码出后续AVFrame */ len1 = avcodec_decode_audio4(is->audio_st->codec, is->audio_frame, &got_frame, pkt); if (len1 < 0) { // error, skip the frame is->audio_pkt_size = 0; break; } is->audio_pkt_data += len1; is->audio_pkt_size -= len1; if (!got_frame) continue; //执行到这里我们得到了一个AVFrame /* 计算解码出来的桢需要的缓冲大小 */ decoded_data_size = av_samples_get_buffer_size(NULL, is->audio_frame->channels, is->audio_frame->nb_samples, is->audio_frame->format, 1); dec_channel_layout = (is->audio_frame->channel_layout && is->audio_frame->channels == av_get_channel_layout_nb_channels( is->audio_frame->channel_layout)) ? is->audio_frame->channel_layout : av_get_default_channel_layout( is->audio_frame->channels); //这个AVFrame每个声道的采样数 wanted_nb_samples = is->audio_frame->nb_samples; /** * 接下来判断我们之前设置SDL时设置的声音格式(AV_SAMPLE_FMT_S16)，声道布局， * 采样频率，每个AVFrame的每个声道采样数与 * 得到的该AVFrame分别是否相同，如有任意不同，我们就需要swr_convert该AvFrame， * 然后才能符合之前设置好的SDL的需要，才能播放 */ if (is->audio_frame->format != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || is->audio_frame->sample_rate != is->audio_src_freq || (wanted_nb_samples != is->audio_frame->nb_samples && !is->swr_ctx)) { if (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, is->audio_frame->format, is->audio_frame->sample_rate, 0, NULL); if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) { fprintf(stderr, "swr_init() failed\n"); break; } is->audio_src_channel_layout = dec_channel_layout; is->audio_src_channels = is->audio_st->codec->channels;