关于解决用tutorial7教程中的代码打造一款自己的播放器中的声音噪音问题

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对于用FFMPEG2.01和SDL2.01最新的版本来做音频播放器，这篇文章和版本是很有值得参考的价值
这篇文章解决了我在做简易播放器的时候，用tutorial07的代码的时候，声音播放出现杂音的问题
出现杂音的问题原因可以从http://blog.csdn.net/leixiaohua1020/article/details/10528443中找到原因，
因为：
“新版本中使用了最新版本的FFMPEG类库（2014.5.7）。
FFMPEG在新版本中的音频解码方面发生了比较大的变化。
如果将旧版的主程序和新版的类库组合使用的话，会出现听到的都是杂音这一现象。
经过研究发现，新版中avcodec_decode_audio4()解码后输出的音频采样数据格式为AV_SAMPLE_FMT_FLTP（float, planar）
而不再是AV_SAMPLE_FMT_S16（signed 16 bits）。因此无法直接使用SDL进行播放。
最后的解决方法是使用SwrContext对音频采样数据进行转换之后，再进行输出播放，问题就可以得到解决了。”

/*=======================来自：http://www.cnblogs.com/lidabo/p/3701074.html================================*/

FFMPEG + SDL音频播放分析
目录 [hide]

1 抽象流程：
2 关键实现：
2.1 main()函数
2.2 decode_thread()读取文件信息和音频包
2.3 stream_component_open()：设置音频参数和打开设备
2.4 audio_callback(): 回调函数，向SDL缓冲区填充数据
2.5 audio_decode_frame()：解码音频
3 FFMPEG结构体
3.1 channel_layout_map
4 FFMPEG宏定义
4.1 Audio channel convenience macros
5 SDL宏定义
5.1 SDL_AudioSpec format
抽象流程：
设置SDL的音频参数 —-> 打开声音设备，播放静音 —-> ffmpeg读取音频流中数据放入队列 —-> SDL调用用户设置的函数来获取音频数据 —-> 播放音频

SDL内部维护了一个buffer来存放解码后的数据，这个buffer中的数据来源是我们注册的回调函数（audio_callback），
audio_callback调用audio_decode_frame来做具体的音频解码工作，需要引起注意的是：
从流中读取出的一个音频包(avpacket)可能含有多个音频桢(avframe)，所以需要多次调用avcodec_decode_audio4来完成整个包的解码，
解码出来的数据存放在我们自己的缓冲中（audio_buf2）。SDL每一次回调都会引起数据从audio_buf2拷贝到SDL内部缓冲区，
当audio_buf2中的数据大于SDL的缓冲区大小时，需要分多次拷贝。

关键实现：

 main()函数
     int main(int argc, char **argv){
     SDL_Event event; //SDL事件变量
     VideoState *is; // 纪录视频及解码器等信息的大结构体
     is = (VideoState*) av_mallocz(sizeof(VideoState));
     if(argc < ){
     　　fprintf(stderr, "Usage: play <file>\n");
     　　exit();
     }
     av_register_all(); //注册所有ffmpeg的解码器
     /* 初始化SDL，这里只实用了AUDIO，如果有视频，好需要SDL_INIT_VIDEO等等 */
     if(SDL_Init(SDL_INIT_AUDIO)){
     　　fprintf(stderr, "Count not initialize SDL - %s\n", SDL_GetError());
     　　exit();
     }
     is_strlcpy(is->filename, argv[], sizeof(is->filename));
     /* 创建一个SDL线程来做视频解码工作，主线程进入SDL事件循环 */
     is->parse_tid = SDL_CreateThread(decode_thread, is);
     if(!is->parse_tid){
     　　SDL_WaitEvent(&event);
     　　switch(event.type){
     　　　　case FF_QUIT_EVENT:
     　　　　case SDL_QUIT:
     　　　　　　is->quit = ;
     　　　　　　SDL_Quit();
     　　　　　　exit();
    　　　　　　 break;
     　　　　default:
    　　　　　　 break;
    　　　 }
     }
     return ;
     }

decode_thread()读取文件信息和音频包：

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

     is->audioStream = -;
     global_video_state = is;
     /* 使用ffmpeg打开视频，解码器等 常规工作 */
     if(avFormat_open_input(&ic, is->filename, NULL, NULL) != ) {
     fprintf(stderr, "open file error: %s\n", is->filename);
     return -;
     }
     is->ic = ic;
     if(avformat_find_stream_info(ic, NULL) < ){
     fprintf(stderr, "find stream info error\n");
     return -;
     }
     av_dump_format(ic, , is->filename, );
     for(i = ; i < ic->nb_streams; i++){
     if(ic->streams[i])->codec->codec_type == AVMEDIA_TYPE_AUDIO && audio_index == -){
     audio_index = i;
     break;
     }
     }
     if(audio_index >= ) {
     /* 所有设置SDL音频流信息的步骤都在这个函数里完成 */
     stream_component_open(is, audio_index);
     }
     if(is->audioStream < ){
     fprintf(stderr, "could not open codecs for file: %s\n", is->filename);
     goto fail;
     }
     /* 读包的主循环， av_read_frame不停的从文件中读取数据包（这里只取音频包）*/
     for(;;){
     if(is->quit) break;
     /* 这里audioq.size是指队列中的所有数据包带的音频数据的总量，并不是包的数量 */
     if(is->audioq.size > MAX_AUDIO_SIZE){
     SDL_Delay(); // 毫秒
     continue;
     }
     ret = av_read_frame(is->ic, packet);
     if(ret < ){
     if(ret == AVERROR_EOF || url_feof(is->ic->pb)) break;
     if(is->ic->pb && is->ic->pb->error) break;
     contiue;
     }
     if(packet->stream_index == is->audioStream){
     packet_queue_put(&is->audioq, packet);
     } else{
     av_free_packet(packet);
     }
     }
     while(!is->quit) SDL_Delay();
     fail: {
     SDL_Event event;
     event.type = FF_QUIT_EVENT;
     event.user.data1 = is;
     SDL_PushEvent(&event);
     }
     return ;
     }

stream_component_open()：设置音频参数和打开设备

     int stream_component_open(videoState *is, int stream_index){
     AVFormatContext *ic = is->ic;
     AVCodecContext *codecCtx;
     AVCodec *codec;
     /* 在用SDL_OpenAudio()打开音频设备的时候需要这两个参数*/
     /* wanted_spec是我们期望设置的属性，spec是系统最终接受的参数 */
     /* 我们需要检查系统接受的参数是否正确 */
     SDL_AudioSpec wanted_spec, spec;
     int64_t wanted_channel_layout = ; // 声道布局(SDL中的具体定义见“FFMPEG结构体”部分)
     int wanted_nb_channels; // 声道数
     /* SDL支持的声道数为 1, 2, 4, 6 */
     /* 后面我们会使用这个数组来纠正不支持的声道数目 */
     const int next_nb_channels[] = { , , , , , , ,  }; 

     if(stream_index <  || stream_index >= ic->nb_streams) return -;
     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_lauout(wanted_channel_nb_channels);
     wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
     }
     wanted_spec.channels = av_get_channels_layout_nb_channels(wanted_channel_layout);
     wanted_spec.freq = codecCtx->sample_rate;
     if(wanted_spec.freq <=  || wanted_spec.channels <=){
     fprintf(stderr, "Invaild sample rate or channel count!\n");
     return -;
     }
     wanted_spec.format = AUDIO_S16SYS; // 具体含义请查看“SDL宏定义”部分
     wanted_spec.silence = ; // 0指示静音
     wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE; // 自定义SDL缓冲区大小
     wanted_spec.callback = audio_callback; // 音频解码的关键回调函数
     wanted_spec.userdata = is; // 传给上面回调函数的外带数据

     /* 打开音频设备，这里使用一个while来循环尝试打开不同的声道数(由上面 */
     /* next_nb_channels数组指定）直到成功打开，或者全部失败 */
     while(SDL_OpenAudio(&wanted_spec, &spec) < ){
     fprintf(stderr, "SDL_OpenAudio(%d channels): %s\n", wanted_spec.channels, SDL_GetError());
     wanted_spec.channels = next_nb_channels[FFMIN(, wanted_spec.channels)]; // FFMIN()由ffmpeg定义的宏，返回较小的数
     if(!wanted_spec.channels){
     fprintf(stderr, "No more channel to try\n");
     return -;
     }
     wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
     }
     /* 检查实际使用的配置（保存在spec,由SDL_OpenAudio()填充） */
     if(spec.format != AUDIO_S16SYS){
     fprintf(stderr, "SDL advised audio format %d is not supported\n", spec.format);
     return -;
     }
     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 support\n", spec.channels);
     return -;
     }
     }
     /* 把设置好的参数保存到大结构中 */
     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_layout = wanted_channel_layout;
     is->audio_src_channels = is->audio_tat_channels = spec.channels;

     codec = avcodec_find_decoder(codecCtx>codec_id);
     if(!codec || (avcodec_open2(codecCtx, codec, NULL) < )){
     fprintf(stderr, "Unsupported codec!\n");
     return -;
     }
     ic->streams[stream_index]->discard = AVDISCARD_DEFAULT; //具体含义请查看“FFMPEG宏定义”部分
     is->audioStream = stream_index;
     is->audio_st = ic->streams[stream_index];
     is->audio_buf_size = ;
     is->audio_buf_index = ;
     memset(&is->audio_pkt, , sizeof(is->audio_pkt));
     packet_queue_init(&is->audioq);
     SDL_PauseAudio(); // 开始播放静音
     }

audio_callback(): 回调函数，向SDL缓冲区填充数据

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

     /* len是由SDL传入的SDL缓冲区的大小，如果这个缓冲未满，我们就一直往里填充数据 */
     while(len > ){
     /* audio_buf_index 和 audio_buf_size 标示我们自己用来放置解码出来的数据的缓冲区，*/
     /* 这些数据待copy到SDL缓冲区， 当audio_buf_index >= audio_buf_size的时候意味着我*/
     /* 们的缓冲为空，没有数据可供copy，这时候需要调用audio_decode_frame来解码出更
     /* 多的桢数据 */
     if(is->audio_buf_index >= is->audio_buf_size){
         audio_data_size = audio_decode_frame(is);
         /* audio_data_size < 0 标示没能解码出数据，我们默认播放静音 */
         if(audio_data_size < )
         {
             is->audio_buf_size = ;
             /* 清零，静音 */
             memset(is->audio_buf, , is->audio_buf_size);
         }
         else
         {
             is->audio_buf_size = audio_data_size;
         }
         is->audio_buf_index = ;
     }
     /* 查看stream可用空间，决定一次copy多少数据，剩下的下次继续copy */
     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;
     }
 }

 audio_decode_frame()：解码音频
     int audio_decode_frame(VideoState *is)
     {
         int len1, len2, decoded_data_size;
         AVPacket *pkt = &is->audio_pkt;
         int got_frame = ;
         int64_t dec_channel_layout;
         int wanted_nb_samples, resampled_data_size;

         for(;;)
         {
         while(is->audio_pkt_size > )
         {
             if(!is->audio_frame)
             {
                 if(!(is->audio_frame = avacodec_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);
             /* 解码错误，跳过整个包 */
             if(len1 < )
             {
                 is->audio_pkt_size = ;
                 break;
             }
             is->audio_pkt_data += len1;
             is->audio_pkt_size -= len1;
             if(!got_frame) continue;
             /* 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, );
             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;
             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,
                                                 , NULL);
                 if (!is->swr_ctx || swr_init(is->swr_ctx) < )
                 {
                     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;
             }
             /* 这里我们可以对采样数进行调整，增加或者减少，一般可以用来做声画同步 */
             if (is->swr_ctx)
             {
                 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) < )
                     {
                         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 < )
                 {
                     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[];
             }
             /* 返回得到的数据 */
             return resampled_data_size;
         }
         if (pkt->data) av_free_packet(pkt);
         memset(pkt, , sizeof(*pkt));
         if (is->quit) return -;
         if (packet_queue_get(&is->audioq, pkt, ) < ) return -;
         is->audio_pkt_data = pkt->data;
         is->audio_pkt_size = pkt->size;

         }
     }

FFMPEG结构体
channel_layout_map
1 static const struct {
2 const char *name;
3 int nb_channels;
4 uint64_t layout;
5 } channel_layout_map[] = {
6 { "mono", 1, AV_CH_LAYOUT_MONO },
7 { "stereo", 2, AV_CH_LAYOUT_STEREO },
8 { "2.1", 3, AV_CH_LAYOUT_2POINT1 },
9 { "3.0", 3, AV_CH_LAYOUT_SURROUND },
10 { "3.0(back)", 3, AV_CH_LAYOUT_2_1 },
11 { "4.0", 4, AV_CH_LAYOUT_4POINT0 },
12 { "quad", 4, AV_CH_LAYOUT_QUAD },
13 { "quad(side)", 4, AV_CH_LAYOUT_2_2 },
14 { "3.1", 4, AV_CH_LAYOUT_3POINT1 },
15 { "5.0", 5, AV_CH_LAYOUT_5POINT0_BACK },
16 { "5.0(side)", 5, AV_CH_LAYOUT_5POINT0 },
17 { "4.1", 5, AV_CH_LAYOUT_4POINT1 },
18 { "5.1", 6, AV_CH_LAYOUT_5POINT1_BACK },
19 { "5.1(side)", 6, AV_CH_LAYOUT_5POINT1 },
20 { "6.0", 6, AV_CH_LAYOUT_6POINT0 },
21 { "6.0(front)", 6, AV_CH_LAYOUT_6POINT0_FRONT },
22 { "hexagonal", 6, AV_CH_LAYOUT_HEXAGONAL },
23 { "6.1", 7, AV_CH_LAYOUT_6POINT1 },
24 { "6.1", 7, AV_CH_LAYOUT_6POINT1_BACK },
25 { "6.1(front)", 7, AV_CH_LAYOUT_6POINT1_FRONT },
26 { "7.0", 7, AV_CH_LAYOUT_7POINT0 },
27 { "7.0(front)", 7, AV_CH_LAYOUT_7POINT0_FRONT },
28 { "7.1", 8, AV_CH_LAYOUT_7POINT1 },
29 { "7.1(wide)", 8, AV_CH_LAYOUT_7POINT1_WIDE },
30 { "octagonal", 8, AV_CH_LAYOUT_OCTAGONAL },
31 { "downmix", 2, AV_CH_LAYOUT_STEREO_DOWNMIX, },
32 };

FFMPEG宏定义
Audio channel convenience macros
1 #define AV_CH_LAYOUT_MONO (AV_CH_FRONT_CENTER)
2 #define AV_CH_LAYOUT_STEREO (AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT)
3 #define AV_CH_LAYOUT_2POINT1 (AV_CH_LAYOUT_STEREO|AV_CH_LOW_FREQUENCY)
4 #define AV_CH_LAYOUT_2_1 (AV_CH_LAYOUT_STEREO|AV_CH_BACK_CENTER)
5 #define AV_CH_LAYOUT_SURROUND (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER)
6 #define AV_CH_LAYOUT_3POINT1 (AV_CH_LAYOUT_SURROUND|AV_CH_LOW_FREQUENCY)
7 #define AV_CH_LAYOUT_4POINT0 (AV_CH_LAYOUT_SURROUND|AV_CH_BACK_CENTER)
8 #define AV_CH_LAYOUT_4POINT1 (AV_CH_LAYOUT_4POINT0|AV_CH_LOW_FREQUENCY)
9 #define AV_CH_LAYOUT_2_2 (AV_CH_LAYOUT_STEREO|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)
10 #define AV_CH_LAYOUT_QUAD (AV_CH_LAYOUT_STEREO|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
11 #define AV_CH_LAYOUT_5POINT0 (AV_CH_LAYOUT_SURROUND|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)
12 #define AV_CH_LAYOUT_5POINT1 (AV_CH_LAYOUT_5POINT0|AV_CH_LOW_FREQUENCY)
13 #define AV_CH_LAYOUT_5POINT0_BACK (AV_CH_LAYOUT_SURROUND|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
14 #define AV_CH_LAYOUT_5POINT1_BACK (AV_CH_LAYOUT_5POINT0_BACK|AV_CH_LOW_FREQUENCY)
15 #define AV_CH_LAYOUT_6POINT0 (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_CENTER)
16 #define AV_CH_LAYOUT_6POINT0_FRONT (AV_CH_LAYOUT_2_2|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
17 #define AV_CH_LAYOUT_HEXAGONAL (AV_CH_LAYOUT_5POINT0_BACK|AV_CH_BACK_CENTER)
18 #define AV_CH_LAYOUT_6POINT1 (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER)
19 #define AV_CH_LAYOUT_6POINT1_BACK (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_BACK_CENTER)
20 #define AV_CH_LAYOUT_6POINT1_FRONT (AV_CH_LAYOUT_6POINT0_FRONT|AV_CH_LOW_FREQUENCY)
21 #define AV_CH_LAYOUT_7POINT0 (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
22 #define AV_CH_LAYOUT_7POINT0_FRONT (AV_CH_LAYOUT_5POINT0|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
23 #define AV_CH_LAYOUT_7POINT1 (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
24 #define AV_CH_LAYOUT_7POINT1_WIDE (AV_CH_LAYOUT_5POINT1|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
25 #define AV_CH_LAYOUT_7POINT1_WIDE_BACK (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)
26 #define AV_CH_LAYOUT_OCTAGONAL (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_LEFT|AV_CH_BACK_CENTER|AV_CH_BACK_RIGHT)
27 #define AV_CH_LAYOUT_STEREO_DOWNMIX (AV_CH_STEREO_LEFT|AV_CH_STEREO_RIGHT)

SDL宏定义
SDL_AudioSpec format
1 AUDIO_U8 Unsigned 8-bit samples
2 AUDIO_S8 Signed 8-bit samples
3 AUDIO_U16LSB Unsigned 16-bit samples, in little-endian byte order
4 AUDIO_S16LSB Signed 16-bit samples, in little-endian byte order
5 AUDIO_U16MSB Unsigned 16-bit samples, in big-endian byte order
6 AUDIO_S16MSB Signed 16-bit samples, in big-endian byte order
7 AUDIO_U16 same as AUDIO_U16LSB (for backwards compatability probably)
8 AUDIO_S16 same as AUDIO_S16LSB (for backwards compatability probably)
9 AUDIO_U16SYS Unsigned 16-bit samples, in system byte order
10 AUDIO_S16SYS Signed 16-bit samples, in system byte order

git clone https://github.com/lnmcc/musicPlayer.git