// tutorial07.c

// A pedagogical video player that really works! Now with seeking features.

//

// This tutorial was written by Stephen Dranger (dranger@gmail.com).

//

// Code based on FFplay, Copyright (c) 2003 Fabrice Bellard,

// and a tutorial by Martin Bohme (boehme@inb.uni-luebeckREMOVETHIS.de)

// Tested on Gentoo, CVS version 5/01/07 compiled with GCC 4.1.1

//

// Use the Makefile to build all the samples.

//

// Run using

// tutorial07 myvideofile.mpg

//

// to play the video.

#include <libavcodec/avcodec.h>

#include <libavformat/avformat.h>

#include <libavformat/avio.h>

#include <libswresample/swresample.h>

#include <libswscale/swscale.h>

#include <libavutil/avstring.h>

#include <libavutil/opt.h>

#include <libavutil/time.h>

#include <SDL.h>

#include <SDL_thread.h>

#ifdef __MINGW32__

#undef main /* Prevents SDL from overriding main() */

#endif

#include <stdio.h>

#include <math.h>

#define SDL_AUDIO_BUFFER_SIZE 1024

#define MAX_AUDIO_FRAME_SIZE 192000

#define MAX_AUDIOQ_SIZE (5 * 16 * 1024)

#define MAX_VIDEOQ_SIZE (5 * 256 * 1024)

#define AV_SYNC_THRESHOLD 0.01

#define AV_NOSYNC_THRESHOLD 10.0

#define SAMPLE_CORRECTION_PERCENT_MAX 10

#define AUDIO_DIFF_AVG_NB 20

#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 DEFAULT_AV_SYNC_TYPE AV_SYNC_VIDEO_MASTER

typedef struct PacketQueue {

AVPacketList *first_pkt, *last_pkt;

int nb_packets;

int size;

SDL_mutex *mutex;

SDL_cond *cond;

} PacketQueue;

typedef struct VideoPicture {

SDL_Overlay *bmp;

int width, height; /* source height & width */

int allocated;

double pts;

} VideoPicture;

typedef struct VideoState {

AVFormatContext *pFormatCtx;

int videoStream, audioStream;

int av_sync_type;

double external_clock; /* external clock base */

int64_t external_clock_time;

int seek_req;

int seek_flags;

int64_t seek_pos;

double audio_clock;

AVStream *audio_st;

PacketQueue audioq;

AVFrame audio_frame;

uint8_t audio_buf[(MAX_AUDIO_FRAME_SIZE * 3) / 2];

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;

double audio_diff_cum; /* used for AV difference average computation */

double audio_diff_avg_coef;

double audio_diff_threshold;

int audio_diff_avg_count;

double frame_timer;

double frame_last_pts;

double frame_last_delay;

double video_clock; ///<pts of last decoded frame / predicted pts of next decoded frame

double video_current_pts; ///<current displayed pts (different from video_clock if frame fifos are used)

int64_t video_current_pts_time; ///<time (av_gettime) at which we updated video_current_pts - used to have running video pts

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 *video_tid;

char filename[1024];

int quit;

AVIOContext *io_context;

struct SwsContext *sws_ctx;

struct SwsContext *sws_ctx_audio;

} VideoState;

enum {

AV_SYNC_AUDIO_MASTER,

AV_SYNC_VIDEO_MASTER,

AV_SYNC_EXTERNAL_MASTER,

};

SDL_Surface *screen;

/* Since we only have one decoding thread, the Big Struct

can be global in case we need it. */

VideoState *global_video_state;

AVPacket flush_pkt;

// 初始化队列

void packet_queue_init(PacketQueue *q) {

memset(q, 0, sizeof(PacketQueue));

q->mutex = SDL_CreateMutex();

q->cond = SDL_CreateCond();

}

//　将数据包放置到队列中

int packet_queue_put(PacketQueue *q, AVPacket *pkt) {

AVPacketList *pkt1;

if(pkt != &flush_pkt && av_dup_packet(pkt) < 0) {

return -1;

}

pkt1 = 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;

}

static void packet_queue_flush(PacketQueue *q) {

AVPacketList *pkt, *pkt1;

SDL_LockMutex(q->mutex);

for(pkt = q->first_pkt; pkt != NULL; pkt = pkt1) {

pkt1 = pkt->next;

av_free_packet(&pkt->pkt);

av_freep(&pkt);

}

q->last_pkt = NULL;

q->first_pkt = NULL;

q->nb_packets = 0;

q->size = 0;

SDL_UnlockMutex(q->mutex);

}

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;

}

double get_video_clock(VideoState *is) {

double delta;

delta = (av_gettime() - is->video_current_pts_time) / 1000000.0;

return is->video_current_pts + delta;

}

double get_external_clock(VideoState *is) {

return av_gettime() / 1000000.0;

}

double get_master_clock(VideoState *is) {

if(is->av_sync_type == AV_SYNC_VIDEO_MASTER) {

return get_video_clock(is);

} else if(is->av_sync_type == AV_SYNC_AUDIO_MASTER) {

return get_audio_clock(is);

} else {

return get_external_clock(is);

}

}

/* Add or subtract samples to get a better sync, return new

audio buffer size */

int synchronize_audio(VideoState *is, short *samples,

int samples_size, double pts) {

int n;

double ref_clock;

n = 2 * is->audio_st->codec->channels;

if(is->av_sync_type != AV_SYNC_AUDIO_MASTER) {

double diff, avg_diff;

int wanted_size, min_size, max_size /*, nb_samples */;

ref_clock = get_master_clock(is);

diff = get_audio_clock(is) - ref_clock;

if(diff < AV_NOSYNC_THRESHOLD) {

// accumulate the diffs

is->audio_diff_cum = diff + is->audio_diff_avg_coef

* is->audio_diff_cum;

if(is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) {

is->audio_diff_avg_count++;

} else {

avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);

if(fabs(avg_diff) >= is->audio_diff_threshold) {

wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n);

min_size = samples_size * ((100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100);

max_size = samples_size * ((100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100);

if(wanted_size < min_size) {

wanted_size = min_size;

} else if (wanted_size > max_size) {

wanted_size = max_size;

}

if(wanted_size < samples_size) {

/* remove samples */

samples_size = wanted_size;

} else if(wanted_size > samples_size) {

uint8_t *samples_end, *q;

int nb;

/* add samples by copying final sample*/

nb = (samples_size - wanted_size);

samples_end = (uint8_t *)samples + samples_size - n;

q = samples_end + n;

while(nb > 0) {

memcpy(q, samples_end, n);

q += n;

nb -= n;

}

samples_size = wanted_size;

}

}

}

} else {

/* difference is TOO big; reset diff stuff */

is->audio_diff_avg_count = 0;

is->audio_diff_cum = 0;

}

}

return samples_size;

}

int decode_frame_from_packet(VideoState *is, AVFrame decoded_frame)

{

int64_t src_ch_layout, dst_ch_layout;

int src_rate, dst_rate;

uint8_t **src_data = NULL, **dst_data = NULL;

int src_nb_channels = 0, dst_nb_channels = 0;

int src_linesize, dst_linesize;

int src_nb_samples, dst_nb_samples, max_dst_nb_samples;

enum AVSampleFormat src_sample_fmt, dst_sample_fmt;

int dst_bufsize;

int ret;

src_nb_samples = decoded_frame.nb_samples;

src_linesize = (int) decoded_frame.linesize;

src_data = decoded_frame.data;

if (decoded_frame.channel_layout == 0) {

decoded_frame.channel_layout = av_get_default_channel_layout(decoded_frame.channels);

}

src_rate = decoded_frame.sample_rate;

dst_rate = decoded_frame.sample_rate;

src_ch_layout = decoded_frame.channel_layout;

dst_ch_layout = decoded_frame.channel_layout;

src_sample_fmt = decoded_frame.format;

dst_sample_fmt = AV_SAMPLE_FMT_S16;

av_opt_set_int(is->sws_ctx_audio, "in_channel_layout", src_ch_layout, 0);

av_opt_set_int(is->sws_ctx_audio, "out_channel_layout", dst_ch_layout, 0);

av_opt_set_int(is->sws_ctx_audio, "in_sample_rate", src_rate, 0);

av_opt_set_int(is->sws_ctx_audio, "out_sample_rate", dst_rate, 0);

av_opt_set_sample_fmt(is->sws_ctx_audio, "in_sample_fmt", src_sample_fmt, 0);

av_opt_set_sample_fmt(is->sws_ctx_audio, "out_sample_fmt", dst_sample_fmt, 0);

/* initialize the resampling context */

if ((ret = swr_init(is->sws_ctx_audio)) < 0) {

fprintf(stderr, "Failed to initialize the resampling context\n");

return -1;

}

/* allocate source and destination samples buffers */

src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);

ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels, src_nb_samples, src_sample_fmt, 0);

if (ret < 0) {

fprintf(stderr, "Could not allocate source samples\n");

return -1;

}

/* compute the number of converted samples: buffering is avoided

* ensuring that the output buffer will contain at least all the

* converted input samples */

max_dst_nb_samples = dst_nb_samples = av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

/* buffer is going to be directly written to a rawaudio file, no alignment */

dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);

ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, dst_nb_samples, dst_sample_fmt, 0);

if (ret < 0) {

fprintf(stderr, "Could not allocate destination samples\n");

return -1;

}

/* compute destination number of samples */

dst_nb_samples = av_rescale_rnd(swr_get_delay(is->sws_ctx_audio, src_rate) + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

/* convert to destination format */

ret = swr_convert(is->sws_ctx_audio, dst_data, dst_nb_samples, (const uint8_t **)decoded_frame.data, src_nb_samples);

if (ret < 0) {

fprintf(stderr, "Error while converting\n");

return -1;

}

dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, ret, dst_sample_fmt, 1);

if (dst_bufsize < 0) {

fprintf(stderr, "Could not get sample buffer size\n");

return -1;

}

memcpy(is->audio_buf, dst_data[0], dst_bufsize);

if (src_data) {

av_freep(&src_data[0]);

}

av_freep(&src_data);

if (dst_data) {

av_freep(&dst_data[0]);

}

av_freep(&dst_data);

return dst_bufsize;

}

int audio_decode_frame(VideoState *is, double *pts_ptr) {

int len1, data_size = 0, n;

AVPacket *pkt = &is->audio_pkt;

double pts;

for(;;) {

while(is->audio_pkt_size > 0) {

int got_frame = 0;

len1 = avcodec_decode_audio4(is->audio_st->codec, &is->audio_frame, &got_frame, pkt);

if(len1 < 0) {

/* if error, skip frame */

is->audio_pkt_size = 0;

break;

}

if (got_frame)

{

if (is->audio_frame.format != AV_SAMPLE_FMT_S16) {

data_size = decode_frame_from_packet(is, is->audio_frame);

} else {

data_size =

av_samples_get_buffer_size

(

NULL,

is->audio_st->codec->channels,

is->audio_frame.nb_samples,

is->audio_st->codec->sample_fmt,

1

);

memcpy(is->audio_buf, is->audio_frame.data[0], data_size);

}

}

is->audio_pkt_data += len1;

is->audio_pkt_size -= len1;

if(data_size <= 0) {

/* No data yet, get more frames */

continue;

}

pts = is->audio_clock;

*pts_ptr = pts;

n = 2 * is->audio_st->codec->channels;

is->audio_clock += (double)data_size /

(double)(n * is->audio_st->codec->sample_rate);

/* We have data, return it and come back for more later */

return data_size;

}

if(pkt->data)

av_free_packet(pkt);

if(is->quit) {

return -1;

}

/* next packet */

if(packet_queue_get(&is->audioq, pkt, 1) < 0) {

return -1;

}

if(pkt->data == flush_pkt.data) {

avcodec_flush_buffers(is->audio_st->codec);

continue;

}

is->audio_pkt_data = pkt->data;

is->audio_pkt_size = pkt->size;

/* if update, update the audio clock w/pts */

if(pkt->pts != AV_NOPTS_VALUE) {

is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;

}

}

}

void audio_callback(void *userdata, Uint8 *stream, int len) {

VideoState *is = (VideoState *)userdata;

int len1, audio_size;

double pts;

while(len > 0) {

if(is->audio_buf_index >= is->audio_buf_size) {

/* We have already sent all our data; get more */

audio_size = audio_decode_frame(is, &pts);

if(audio_size < 0) {

/* If error, output silence */

is->audio_buf_size = 1024;

memset(is->audio_buf, 0, is->audio_buf_size);

} else {

audio_size = synchronize_audio(is, (int16_t *)is->audio_buf,

audio_size, pts);

is->audio_buf_size = audio_size;

}

is->audio_buf_index = 0;

}

len1 = is->audio_buf_size - is->audio_buf_index;

if(len1 > len)

len1 = len;

memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);

len -= len1;

stream += len1;

is->audio_buf_index += len1;

}

}

static Uint32 sdl_refresh_timer_cb(Uint32 interval, void *opaque) {

SDL_Event event;

event.type = FF_REFRESH_EVENT;

event.user.data1 = opaque;

SDL_PushEvent(&event);

return 0; /* 0 means stop timer */

}

/* schedule a video refresh in 'delay' ms */

static void schedule_refresh(VideoState *is, int delay) {

SDL_AddTimer(delay, sdl_refresh_timer_cb, is);

}

void video_display(VideoState *is) {

SDL_Rect rect;

VideoPicture *vp;

//AVPicture pict;

float aspect_ratio;

int w, h, x, y;

//int i;

vp = &is->pictq[is->pictq_rindex];

if(vp->bmp) {

if(is->video_st->codec->sample_aspect_ratio.num == 0) {

aspect_ratio = 0;

} else {

aspect_ratio = av_q2d(is->video_st->codec->sample_aspect_ratio) *

is->video_st->codec->width / is->video_st->codec->height;

}

if(aspect_ratio <= 0.0) {

aspect_ratio = (float)is->video_st->codec->width /

(float)is->video_st->codec->height;

}

h = screen->h;

w = ((int)rint(h * aspect_ratio)) & -3;

if(w > screen->w) {

w = screen->w;

h = ((int)rint(w / aspect_ratio)) & -3;

}

x = (screen->w - w) / 2;

y = (screen->h - h) / 2;

rect.x = x;

rect.y = y;

rect.w = w;

rect.h = h;

SDL_DisplayYUVOverlay(vp->bmp, &rect);

}

}

void video_refresh_timer(void *userdata) {

VideoState *is = (VideoState *)userdata;

VideoPicture *vp;

double actual_delay, delay, sync_threshold, ref_clock, diff;

if(is->video_st) {

if(is->pictq_size == 0) {

schedule_refresh(is, 1);

} else {

vp = &is->pictq[is->pictq_rindex];

is->video_current_pts = vp->pts;

is->video_current_pts_time = av_gettime();

delay = vp->pts - is->frame_last_pts; /* the pts from last time */

if(delay <= 0 || delay >= 1.0) {

/* if incorrect delay, use previous one */

delay = is->frame_last_delay;

}

/* save for next time */

is->frame_last_delay = delay;

is->frame_last_pts = vp->pts;

/* update delay to sync to audio if not master source */

if(is->av_sync_type != AV_SYNC_VIDEO_MASTER) {

ref_clock = get_master_clock(is);

diff = vp->pts - ref_clock;

/* Skip or repeat the frame. Take delay into account

FFPlay still doesn't "know if this is the best guess." */

sync_threshold = (delay > AV_SYNC_THRESHOLD) ? delay : AV_SYNC_THRESHOLD;

if(fabs(diff) < AV_NOSYNC_THRESHOLD) {

if(diff <= -sync_threshold) {

delay = 0;

} else if(diff >= sync_threshold) {

delay = 2 * delay;

}

}

}

is->frame_timer += delay;

/* computer the REAL delay */

actual_delay = is->frame_timer - (av_gettime() / 1000000.0);

if(actual_delay < 0.010) {

/* Really it should skip the picture instead */

actual_delay = 0.010;

}

schedule_refresh(is, (int)(actual_delay * 1000 + 0.5));

/* show the picture! */

video_display(is);

/* update queue for next picture! */

if(++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE) {

is->pictq_rindex = 0;

}

SDL_LockMutex(is->pictq_mutex);

is->pictq_size--;

SDL_CondSignal(is->pictq_cond);

SDL_UnlockMutex(is->pictq_mutex);

}

} else {

schedule_refresh(is, 100);

}

}

void alloc_picture(void *userdata) {

VideoState *is = (VideoState *)userdata;

VideoPicture *vp;

vp = &is->pictq[is->pictq_windex];

if(vp->bmp) {

// we already have one make another, bigger/smaller

SDL_FreeYUVOverlay(vp->bmp);

}

// Allocate a place to put our YUV image on that screen

vp->bmp = SDL_CreateYUVOverlay(is->video_st->codec->width,

is->video_st->codec->height,

SDL_YV12_OVERLAY,

screen);

vp->width = is->video_st->codec->width;

vp->height = is->video_st->codec->height;

SDL_LockMutex(is->pictq_mutex);

vp->allocated = 1;

SDL_CondSignal(is->pictq_cond);

SDL_UnlockMutex(is->pictq_mutex);

}

int queue_picture(VideoState *is, AVFrame *pFrame, double pts) {

VideoPicture *vp;

AVPicture pict;

/* wait until we have space for a new pic */

SDL_LockMutex(is->pictq_mutex);

while(is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE &&

!is->quit) {

SDL_CondWait(is->pictq_cond, is->pictq_mutex);

}

SDL_UnlockMutex(is->pictq_mutex);

if(is->quit)

return -1;

// windex is set to 0 initially

vp = &is->pictq[is->pictq_windex];

/* allocate or resize the buffer! */

if(!vp->bmp ||

vp->width != is->video_st->codec->width ||

vp->height != is->video_st->codec->height) {

SDL_Event event;

vp->allocated = 0;

/* we have to do it in the main thread */

event.type = FF_ALLOC_EVENT;

event.user.data1 = is;

SDL_PushEvent(&event);

/* wait until we have a picture allocated */

SDL_LockMutex(is->pictq_mutex);

while(!vp->allocated && !is->quit) {

SDL_CondWait(is->pictq_cond, is->pictq_mutex);

}

SDL_UnlockMutex(is->pictq_mutex);

if(is->quit) {

return -1;

}

}

/* We have a place to put our picture on the queue */

/* If we are skipping a frame, do we set this to null

but still return vp->allocated = 1? */

if(vp->bmp) {

SDL_LockYUVOverlay(vp->bmp);

/* point pict at the queue */

pict.data[0] = vp->bmp->pixels[0];

pict.data[1] = vp->bmp->pixels[2];

pict.data[2] = vp->bmp->pixels[1];

pict.linesize[0] = vp->bmp->pitches[0];

pict.linesize[1] = vp->bmp->pitches[2];

pict.linesize[2] = vp->bmp->pitches[1];

// Convert the image into YUV format that SDL uses

sws_scale

(

is->sws_ctx,

(uint8_t const * const *)pFrame->data,

pFrame->linesize,

0,

is->video_st->codec->height,

pict.data,

pict.linesize

);

SDL_UnlockYUVOverlay(vp->bmp);

vp->pts = pts;

/* now we inform our display thread that we have a pic ready */

if(++is->pictq_windex == VIDEO_PICTURE_QUEUE_SIZE) {

is->pictq_windex = 0;

}

SDL_LockMutex(is->pictq_mutex);

is->pictq_size++;

SDL_UnlockMutex(is->pictq_mutex);

}

return 0;

}

double synchronize_video(VideoState *is, AVFrame *src_frame, double pts) {

double frame_delay;

if(pts != 0) {

/* if we have pts, set video clock to it */

is->video_clock = pts;

} else {

/* if we aren't given a pts, set it to the clock */

pts = is->video_clock;

}

/* update the video clock */

frame_delay = av_q2d(is->video_st->codec->time_base);

/* if we are repeating a frame, adjust clock accordingly */

frame_delay += src_frame->repeat_pict * (frame_delay * 0.5);

is->video_clock += frame_delay;

return pts;

}

uint64_t global_video_pkt_pts = AV_NOPTS_VALUE;

/* These are called whenever we allocate a frame

* buffer. We use this to store the global_pts in

* a frame at the time it is allocated.

*/

int our_get_buffer(struct AVCodecContext *c, AVFrame *pic) {

int ret = avcodec_default_get_buffer(c, pic);

uint64_t *pts = av_malloc(sizeof(uint64_t));

*pts = global_video_pkt_pts;

pic->opaque = pts;

return ret;

}

void our_release_buffer(struct AVCodecContext *c, AVFrame *pic) {

if(pic) av_freep(&pic->opaque);

avcodec_default_release_buffer(c, pic);

}

int video_thread(void *arg) {

VideoState *is = (VideoState *)arg;

AVPacket pkt1, *packet = &pkt1;

int frameFinished;

AVFrame *pFrame;

double pts;

pFrame = av_frame_alloc();

for(;;) {

if(packet_queue_get(&is->videoq, packet, 1) < 0) {

// means we quit getting packets

break;

}

if(packet->data == flush_pkt.data) {

avcodec_flush_buffers(is->video_st->codec);

continue;

}

pts = 0;

// Save global pts to be stored in pFrame in first call

global_video_pkt_pts = packet->pts;

// Decode video frame

avcodec_decode_video2(is->video_st->codec, pFrame, &frameFinished,

packet);

if(packet->dts == AV_NOPTS_VALUE

&& pFrame->opaque && *(uint64_t*)pFrame->opaque != AV_NOPTS_VALUE) {

pts = *(uint64_t *)pFrame->opaque;

} else if(packet->dts != AV_NOPTS_VALUE) {

pts = packet->dts;

} else {

pts = 0;

}

pts *= av_q2d(is->video_st->time_base);

// Did we get a video frame?

if(frameFinished) {

pts = synchronize_video(is, pFrame, pts);

if(queue_picture(is, pFrame, pts) < 0) {

break;

}

}

av_free_packet(packet);

}

av_free(pFrame);

return 0;

}

int stream_component_open(VideoState *is, int stream_index) {

AVFormatContext *pFormatCtx = is->pFormatCtx;

AVCodecContext *codecCtx = NULL;

AVCodec *codec = NULL;

AVDictionary *optionsDict = NULL;

SDL_AudioSpec wanted_spec, spec;

if(stream_index < 0 || stream_index >= pFormatCtx->nb_streams) {

return -1;

}

// Get a pointer to the codec context for the video stream

codecCtx = pFormatCtx->streams[stream_index]->codec;

if(codecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {

// Set audio settings from codec info

wanted_spec.freq = codecCtx->sample_rate;

wanted_spec.format = AUDIO_S16SYS;

wanted_spec.channels = codecCtx->channels;

wanted_spec.silence = 0;

wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;

wanted_spec.callback = audio_callback;

wanted_spec.userdata = is;

if(SDL_OpenAudio(&wanted_spec, &spec) < 0) {

fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError());

return -1;

}

is->audio_hw_buf_size = spec.size;

}

codec = avcodec_find_decoder(codecCtx->codec_id);

if(!codec || (avcodec_open2(codecCtx, codec, &optionsDict) < 0)) {

fprintf(stderr, "Unsupported codec!\n");

return -1;

}

switch(codecCtx->codec_type) {

case AVMEDIA_TYPE_AUDIO:

is->audioStream = stream_index;

is->audio_st = pFormatCtx->streams[stream_index];

is->audio_buf_size = 0;

is->audio_buf_index = 0;

/* averaging filter for audio sync */

is->audio_diff_avg_coef = exp(log(0.01 / AUDIO_DIFF_AVG_NB));

is->audio_diff_avg_count = 0;

/* Correct audio only if larger error than this */

is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / codecCtx->sample_rate;

is->sws_ctx_audio = swr_alloc();

if (!is->sws_ctx_audio) {

fprintf(stderr, "Could not allocate resampler context\n");

return -1;

}

memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));

packet_queue_init(&is->audioq);

SDL_PauseAudio(0);

break;

case AVMEDIA_TYPE_VIDEO:

is->videoStream = stream_index;

is->video_st = pFormatCtx->streams[stream_index];

is->frame_timer = (double)av_gettime() / 1000000.0;

is->frame_last_delay = 40e-3;

is->video_current_pts_time = av_gettime();

packet_queue_init(&is->videoq);

is->video_tid = SDL_CreateThread(video_thread, is);

is->sws_ctx =

sws_getContext

(

is->video_st->codec->width,

is->video_st->codec->height,

is->video_st->codec->pix_fmt,

is->video_st->codec->width,

is->video_st->codec->height,

AV_PIX_FMT_YUV420P,

SWS_BILINEAR,

NULL,

NULL,

NULL

);

codecCtx->get_buffer2 = our_get_buffer;

codecCtx->release_buffer = our_release_buffer;

break;

default:

break;

}

return 0;

}

int decode_interrupt_cb(void *opaque) {

return (global_video_state && global_video_state->quit);

}

int decode_thread(void *arg) {

// 上半部分的函数没什么新东西；它的工作就是打开文件和找到视频流和音频流的索引。

// 唯一不同的地方是把格式内容保存到大结构体中。当找到流后，调用另一个将要定义的函数 stream_component_open()。

// 这是一个一般的分离的方法，自从我们设置很多相似的视频和音频解码的代码，我们通过编写这个函数来重用它们。

VideoState *is = (VideoState *)arg;

AVFormatContext *pFormatCtx = NULL;

AVPacket pkt1, *packet = &pkt1;

// http://www.ffmpeg.org/doxygen/3.1/group__lavu__dict.html#details

AVDictionary *io_dict = NULL; // AVDictionary 元数据,Simple key:value store.

// http://www.ffmpeg.org/doxygen/3.1/structAVIOInterruptCB.html#details

AVIOInterruptCB callback; // Callback for checking whether to abort blocking functions.

int video_index = -1;

int audio_index = -1;

int i;

is->videoStream=-1;

is->audioStream=-1;

global_video_state = is;

// will interrupt blocking functions if we quit!

callback.callback = decode_interrupt_cb;

callback.opaque = is;

// http://www.ffmpeg.org/doxygen/3.1/aviobuf_8c.html#ae8589aae955d16ca228b6b9d66ced33d

// 使用is->filename的内容初始化is->io_context，用于管理文件的读写

if (avio_open2(&is->io_context, is->filename, 0, &callback, &io_dict))

{

fprintf(stderr, "Unable to open I/O for %s\n", is->filename);

return -1;

}

// 从传入的第二个参数获得文件路径，这个函数会读取文件头信息，并把信息保存在 pFormatCtx 结构体当中。

// 这个函数后面两个参数分别是： 指定文件格式、格式化选项，当我们设置为 NULL 或 0 时，libavformat 会自动完成这些工作。

if(avformat_open_input(&pFormatCtx, is->filename, NULL, NULL)!=0)

return -1; // Couldn't open file

is->pFormatCtx = pFormatCtx;

// Retrieve stream information

// 得到流信息

if(avformat_find_stream_info(pFormatCtx, NULL)<0)

return -1; // Couldn't find stream information

// Dump information about file onto standard error

// 这个函数填充了 pFormatCtx->streams 流信息， 可以通过 dump_format 把信息打印出来：

av_dump_format(pFormatCtx, 0, is->filename, 0);

// 查找一个音频和视频流

for(i=0; i<pFormatCtx->nb_streams; i++) {

if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO &&

video_index < 0) {

video_index=i;

}

if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &&

audio_index < 0) {

audio_index=i;

}

}

// stream_component_open()函数的作用是找到解码器，设置音频参数，保存重要信息到大结构体中，然后启动音频和视频线程。

if(audio_index >= 0) {

stream_component_open(is, audio_index);

}

if(video_index >= 0) {

stream_component_open(is, video_index);

}

if(is->videoStream < 0 || is->audioStream < 0) {

fprintf(stderr, "%s: could not open codecs\n", is->filename);

goto fail;

}

// main decode loop

// 上面都是打开文件和找到视频流和音频流的索引等工作，下面才是主循环

for(;;) {

if(is->quit) {

break; //　控制循环是否退出

}

// seek stuff goes here

if(is->seek_req) {

int stream_index= -1;

int64_t seek_target = is->seek_pos;

if(is->videoStream >= 0) stream_index = is->videoStream;

else if(is->audioStream >= 0) stream_index = is->audioStream;

if(stream_index>=0){

seek_target= av_rescale_q(seek_target, AV_TIME_BASE_Q, pFormatCtx->streams[stream_index]->time_base);

}

if(av_seek_frame(is->pFormatCtx, stream_index, seek_target, is->seek_flags) < 0) {

fprintf(stderr, "%s: error while seeking\n", is->pFormatCtx->filename);

} else {

if(is->audioStream >= 0) {

packet_queue_flush(&is->audioq);

packet_queue_put(&is->audioq, &flush_pkt);

}