linux下ffmpeg sdl开发

1、确认Makefile中指定的config.mak（在ffmpeg根目录下）中：CONFIG_FFPLAY=yes，如果不是需要重新./configure

2、编译时需要安装libsdl1.2-dev，命令为sudo apt-get install libsdl1.2-dev，如果安装失败，之前如果安装过sdl-devel包最好将其卸载。根据错误原因来解决。

3、make，结束后发现ffplay已经生成了

4、make install

一直想弄个音乐播放器来着,现有的播放器框架很多比如Qt就有phono框架,但是感觉框架类结构很大让人抓不到头脑,于是就想从底层做起,好吧还是说大话了,本文决定使用ffmpeg+sdl来着手

到ffmpeg官网上下载源码编译(PS: 如果要生成ffplay要先安装sdl 1.2)
命令行安装sdl 1.2
编译安装 ./configure make make install


完成后要解决怎么使用的问题了,下面无耻的贴出网上搜罗来的代码
编译时可能会报错,贴上大婶的Makefile,其实我想说的是这个makefile还是让人蛋疼的选项太多了
到ffmpeg源码目录下运行 find ./ -name "*.pc"输出如下
./libavcodec/libavcodec.pc
./libavdevice/libavdevice.pc
./libswresample/libswresample.pc
./doc/examples/pc-uninstalled/libswresample.pc
./doc/examples/pc-uninstalled/libavdevice.pc
./doc/examples/pc-uninstalled/libswscale.pc
./doc/examples/pc-uninstalled/libavutil.pc
./doc/examples/pc-uninstalled/libavcodec.pc
./doc/examples/pc-uninstalled/libavformat.pc
./doc/examples/pc-uninstalled/libavfilter.pc
./libavutil/libavutil.pc
./libavformat/libavformat.pc
./libswscale/libswscale.pc
./libavfilter/libavfilter.pc


上面的pc文件就是运行pkg-config --libs 时候用到的了
但是太多的选项于是想自己动手
在 /usr/local/lib/pkgconfig/ 目录下建立 ffmpeg.pc内容如下:(一看就是偷懒复制libavutil.pc得来的,主要是LIbs:话说该输出是使用大婶Makefile中的pkg-config --libs -lavdevice  -lswresample -lavformat -lavcodec -lavutil -lswscale 得到的)
prefix=/usr/local
exec_prefix=${prefix}
libdir=${prefix}/lib
includedir=${prefix}/include


Name: libavutil
Description: FFmpeg utility library
Version: 54.15.100
Requires: 
Requires.private: 
Conflicts:
Libs: -L${libdir}  -pthread -L/usr/local/lib -lavdevice -lavfilter -lswscale -lavformat -lavcodec -lxcb-shm -lxcb -lX11 -lasound -lSDL -lz -lswresample -lavutil -lrt -lm
Libs.private: 
Cflags: -I${includedir}


这样以后运行pkg-config --libs ffmpeg就可以编译ffmpeg程序了


贴上Makefile.am,话说怎么使用GNU的auto系列工具见前一篇文章
###################################################################################
bin_PROGRAMS=musicPlayer
musicPlayer_SOURCES=main.c
LIBS=`sdl-config --libs`
INCLUDE=`sdl-config --cflags`


LIBS+=`pkg-config --libs ffmpeg`


######################################################################




TARGET = player




CROSS_COMPILE =


#####################################
AS              = $(CROSS_COMPILE)as
LD              = $(CROSS_COMPILE)ld
CC              = $(CROSS_COMPILE)gcc
CPP             = $(CC) -E
AR              = $(CROSS_COMPILE)ar
NM              = $(CROSS_COMPILE)nm
STRIP           = $(CROSS_COMPILE)strip
OBJCOPY         = $(CROSS_COMPILE)objcopy
OBJDUMP         = $(CROSS_COMPILE)objdump
LN = ln -f
CHMOD = chmod


CFLAGS += -I. -I/usr/local/include -g -Wall  -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_ISOC9X_SOURCE -std=c99


LDFLAGS += -lavdevice  -lswresample -lavformat -lavcodec -lavutil -lswscale -lGLU -lGL -lm -lz -lpthread -lX11 -lSDL


SRC=$(wildcard *.c)
OBJS=${SRC:%.c=%.o}
NAME=${SRC:%.c=%}
DEPS=$(SRC:%.c=.dep/%.d)


.PHONY: dep  all


all: $(OBJS)
$(CC) -o $(TARGET) $(OBJS) $(LDFLAGS) 
# $(STRIP) $(TARGET)




clean:
rm -rf *.o $(TARGET) .dep


%.o: %.c
${CC} ${CFLAGS} -c $<
@mkdir -p .dep
${CC} -MM $(CFLAGS) $*.c > .dep/$*.d 




dep: 
@mkdir -p .dep
for i in ${NAME} ; do  \
${CC} -MM $(CFLAGS) "$${i}".c > .dep/"$${i}".d ;\
done

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#include <libavutil/avstring.h>
#include <libavutil/pixfmt.h>
#include <libavutil/log.h>
#include <SDL/SDL.h>
#include <SDL/SDL_thread.h>
#include <stdio.h>
#include <math.h>


#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
#define SDL_AUDIO_BUFFER_SIZE 1024 
#define MAX_AUDIOQ_SIZE (1 * 1024 * 1024)
#define FF_ALLOC_EVENT   (SDL_USEREVENT)
#define FF_REFRESH_EVENT (SDL_USEREVENT + 1)
#define FF_QUIT_EVENT (SDL_USEREVENT + 2)


typedef struct PacketQueue {
    AVPacketList *first_pkt, *last_pkt;
    int nb_packets;
    int size;
    SDL_mutex *mutex;
    SDL_cond *cond;
} PacketQueue;


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;
    uint8_t         *audio_buf;
    uint8_t         *audio_buf1;
    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;
    SDL_Thread      *parse_tid;
    int             quit;
} VideoState;


VideoState *global_video_state;


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;
}


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);
}


int audio_decode_frame(VideoState *is) {
    int len1, len2, decoded_data_size;
    AVPacket *pkt = &is->audio_pkt;
    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 = avcodec_alloc_frame())) {
                    return AVERROR(ENOMEM);
                }
            } else 
                avcodec_get_frame_defaults(is->audio_frame);


            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;


            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);


            wanted_nb_samples =  is->audio_frame->nb_samples;


            //fprintf(stderr, "wanted_nb_samples = %d\n", wanted_nb_samples);


            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;
                is->audio_src_freq = is->audio_st->codec->sample_rate;
                is->audio_src_fmt = is->audio_st->codec->sample_fmt;
            }
            if (is->swr_ctx) {
               // const uint8_t *in[] = { is->audio_frame->data[0] };
                const uint8_t **in = (const uint8_t **)is->audio_frame->extended_data; 
                uint8_t *out[] = { is->audio_buf2 };
				if (wanted_nb_samples != is->audio_frame->nb_samples) {
					 if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->audio_frame->nb_samples)
												 * is->audio_tgt_freq / is->audio_frame->sample_rate,
												 wanted_nb_samples * is->audio_tgt_freq / is->audio_frame->sample_rate) < 0) {
						 fprintf(stderr, "swr_set_compensation() failed\n");
						 break;
					 }
				 }


                len2 = swr_convert(is->swr_ctx, out,
                                   sizeof(is->audio_buf2)
                                   / is->audio_tgt_channels
                                   / av_get_bytes_per_sample(is->audio_tgt_fmt),
                                   in, is->audio_frame->nb_samples);
                if (len2 < 0) {
                    fprintf(stderr, "swr_convert() failed\n");
                    break;
                }
                if (len2 == sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt)) {
                    fprintf(stderr, "warning: audio buffer is probably too small\n");
                    swr_init(is->swr_ctx);
                }
                is->audio_buf = is->audio_buf2;
                resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
            } else {
				resampled_data_size = decoded_data_size;
                is->audio_buf = is->audio_frame->data[0];
            }
            // We have data, return it and come back for more later
            return resampled_data_size;
        }


        if (pkt->data) av_free_packet(pkt);
		memset(pkt, 0, sizeof(*pkt));
        if (is->quit) return -1;
        if (packet_queue_get(&is->audioq, pkt, 1) < 0) return -1;


        is->audio_pkt_data = pkt->data;
        is->audio_pkt_size = pkt->size;
    }
}


void audio_callback(void *userdata, Uint8 *stream, int len) {
    VideoState *is = (VideoState *)userdata;
    int len1, audio_data_size;


    while (len > 0) {
        if (is->audio_buf_index >= is->audio_buf_size) {
            audio_data_size = audio_decode_frame(is);


            if(audio_data_size < 0) {
                /* silence */
                is->audio_buf_size = 1024;
                memset(is->audio_buf, 0, is->audio_buf_size);
            } else {
                is->audio_buf_size = audio_data_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;
    }
}


int stream_component_open(VideoState *is, int stream_index) {
    AVFormatContext *ic = is->ic;
    AVCodecContext *codecCtx;
    AVCodec *codec;
    SDL_AudioSpec wanted_spec, spec;
    int64_t wanted_channel_layout = 0;
    int wanted_nb_channels;
	const int next_nb_channels[] = {0, 0, 1 ,6, 2, 6, 4, 6};


    if (stream_index < 0 || stream_index >= ic->nb_streams) {
        return -1;
    }
	
    codecCtx = ic->streams[stream_index]->codec;
	wanted_nb_channels = codecCtx->channels;
	if(!wanted_channel_layout || wanted_nb_channels != av_get_channel_layout_nb_channels(wanted_channel_layout)) {
		wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
		wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
	}
	
	wanted_spec.channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
	wanted_spec.freq = codecCtx->sample_rate;
	if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
		fprintf(stderr, "Invalid sample rate or channel count!\n");
		return -1;
	}
	wanted_spec.format = AUDIO_S16SYS;
	wanted_spec.silence = 0;
	wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
	wanted_spec.callback = audio_callback;
	wanted_spec.userdata = is;
	
	while(SDL_OpenAudio(&wanted_spec, &spec) < 0) {
		fprintf(stderr, "SDL_OpenAudio (%d channels): %s\n", wanted_spec.channels, SDL_GetError());
		wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
		if(!wanted_spec.channels) {
			fprintf(stderr, "No more channel combinations to tyu, audio open failed\n");
			return -1;
		}
		wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
	}


	if (spec.format != AUDIO_S16SYS) {
		fprintf(stderr, "SDL advised audio format %d is not supported!\n", spec.format);
		return -1;
	}
	if (spec.channels != wanted_spec.channels) {
		wanted_channel_layout = av_get_default_channel_layout(spec.channels);
		if (!wanted_channel_layout) {
			fprintf(stderr, "SDL advised channel count %d is not supported!\n", spec.channels);
			return -1;
		}
	}


	fprintf(stderr, "%d: wanted_spec.format = %d\n", __LINE__, wanted_spec.format);
	fprintf(stderr, "%d: wanted_spec.samples = %d\n", __LINE__, wanted_spec.samples);
	fprintf(stderr, "%d: wanted_spec.channels = %d\n", __LINE__, wanted_spec.channels);
	fprintf(stderr, "%d: wanted_spec.freq = %d\n", __LINE__, wanted_spec.freq);


	fprintf(stderr, "%d: spec.format = %d\n", __LINE__, spec.format);
	fprintf(stderr, "%d: spec.samples = %d\n", __LINE__, spec.samples);
	fprintf(stderr, "%d: spec.channels = %d\n", __LINE__, spec.channels);
	fprintf(stderr, "%d: spec.freq = %d\n", __LINE__, spec.freq);


	is->audio_src_fmt = is->audio_tgt_fmt = AV_SAMPLE_FMT_S16;
	is->audio_src_freq = is->audio_tgt_freq = spec.freq;
	is->audio_src_channel_layout = is->audio_tgt_channel_layout = wanted_channel_layout;
	is->audio_src_channels = is->audio_tgt_channels = spec.channels;
    
    codec = avcodec_find_decoder(codecCtx->codec_id);
    if (!codec || (avcodec_open2(codecCtx, codec, NULL) < 0)) {
        fprintf(stderr, "Unsupported codec!\n");
        return -1;
    }
	ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
    switch(codecCtx->codec_type) {
    case AVMEDIA_TYPE_AUDIO:
        is->audioStream = stream_index;
        is->audio_st = ic->streams[stream_index];
        is->audio_buf_size = 0;
        is->audio_buf_index = 0;
        memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
        packet_queue_init(&is->audioq);
        SDL_PauseAudio(0);
        break;
    default:
        break;
    }
}
/*
static void stream_component_close(VideoState *is, int stream_index) {
	AVFormatContext *oc = is->;
	AVCodecContext *avctx;


	if(stream_index < 0 || stream_index >= ic->nb_streams)	return;
	avctx = ic->streams[stream_index]->codec;


}
*/
static int decode_thread(void *arg) {
    VideoState *is = (VideoState *)arg;
    AVFormatContext *ic = NULL;
    AVPacket pkt1, *packet = &pkt1;
    int ret, i, audio_index = -1;


    is->audioStream=-1;
    global_video_state = is;
    if (avformat_open_input(&ic, is->filename, NULL, NULL) != 0) {
        return -1;
    }
    is->ic = ic;
    if (avformat_find_stream_info(ic, NULL) < 0) {
        return -1;
    }
    av_dump_format(ic, 0, is->filename, 0);
    for (i=0; i<ic->nb_streams; i++) {
        if (ic->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO && audio_index < 0) {
            audio_index=i;
            break;
        }
    }
    if (audio_index >= 0) {
        stream_component_open(is, audio_index);
    }
    if (is->audioStream < 0) {
        fprintf(stderr, "%s: could not open codecs\n", is->filename);
        goto fail;
    }
    // main decode loop
    for(;;) {
        if(is->quit) break;
        if (is->audioq.size > MAX_AUDIOQ_SIZE) {
            SDL_Delay(10);
            continue;
        }
        ret = av_read_frame(is->ic, packet);
        if (ret < 0) {
            if(ret == AVERROR_EOF || url_feof(is->ic->pb)) {
                break;
            }
            if(is->ic->pb && is->ic->pb->error) {
                break;
            }
            continue;
        }


        if (packet->stream_index == is->audioStream) {
            packet_queue_put(&is->audioq, packet);
        } else {
            av_free_packet(packet);
        }
    }


    while (!is->quit) {
        SDL_Delay(100);
    }


fail: {
        SDL_Event event;
        event.type = FF_QUIT_EVENT;
        event.user.data1 = is;
        SDL_PushEvent(&event);
    }


    return 0;
}


int main(int argc, char *argv[]) {
    SDL_Event       event;
    VideoState      *is;


    is = (VideoState *)av_mallocz(sizeof(VideoState));


    if (argc < 2) {
        fprintf(stderr, "Usage: test <file>\n");
        exit(1);
    }


    av_register_all();


    if (SDL_Init(SDL_INIT_AUDIO)) {
        fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError());
        exit(1);
    }


    av_strlcpy(is->filename, argv[1], sizeof(is->filename));


    is->parse_tid = SDL_CreateThread(decode_thread, is);
    if (!is->parse_tid) {
        av_free(is);
        return -1;
    }


    for(;;) {
        SDL_WaitEvent(&event);
        switch(event.type) {
        case FF_QUIT_EVENT:
        case SDL_QUIT:
            is->quit = 1;
            SDL_Quit();
            exit(0);
            break;
        default:
            break;
        }
    }
    return 0;
}

其实本文就是想使用pkg-config --libs ffmpeg来编译程序
好吧打完收招,看大婶的程序去了