ffmpeg 版本:git clone 于 2014-12-02 ,版本接近2.44,在2.44和2.51之间
SDL版本:SDL 1.2(Centos 6.5软件库的相应版本)
有些旧的ffmpeg播放音频示例中,会存在一些音频可以播放一些不能播放,其中一个我们需要考虑的原因和该注意的地方就是 av_decode_audiole类似函数所获的的AVFrame的格式是否是我们(SDL)所需要的,本例代码用来解决该问题,关键点在于swr_convert函数,代码及注释如下:
#include#include #include #include #include #include #include #include #include #include #include #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) //该字段存在于旧版本的ffmpeg中,此处粘贴过来使用,勿怪! #define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio 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; } 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 = av_frame_alloc())) { return AVERROR(ENOMEM); } } else av_frame_unref(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 decoded_data_size = av_samples_get_buffer_size(NULL, is->audio_frame->channels, is->audio_frame->nb_samples, is->audio_frame->format, 1); //得到这个AvFrame的声音布局,比如立体声 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; is->audio_src_freq = is->audio_st->codec->sample_rate; is->audio_src_fmt = is->audio_st->codec->sample_fmt; } /** * 如果上面if判断失败,就会初始化好swr_ctx,就会如期进行转换 */ 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) { fprintf(stdout, "swr_set_compensation \n"); 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; } } /** * 转换该AVFrame到设置好的SDL需要的样子,有些旧的代码示例最主要就是少了这一部分, * 往往一些音频能播,一些不能播,这就是原因,比如有些源文件音频恰巧是AV_SAMPLE_FMT_S16的。 * swr_convert 返回的是转换后每个声道(channel)的采样数 */ 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; //每声道采样数 x 声道数 x 每个采样字节数 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; } } /** * 设置SDL播放声音的参数如声音采样格式,声道布局,静音值 */ 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; } } /** * demuxing出AVPacket */ 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 \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; }
FFmpeg版本逐渐更新,代码功能更加丰富和易于使用,掌握音视频基础概念结合ffmpeg就可以方便使用!