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C++ FFMIN函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

本文整理汇总了C++中FFMIN函数的典型用法代码示例。如果您正苦于以下问题:C++ FFMIN函数的具体用法?C++ FFMIN怎么用?C++ FFMIN使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。



在下文中一共展示了FFMIN函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: CheckPointer

STDMETHODIMP CDecAvcodec::Decode(const BYTE *buffer, int buflen, REFERENCE_TIME rtStartIn, REFERENCE_TIME rtStopIn, BOOL bSyncPoint, BOOL bDiscontinuity)
{
  CheckPointer(m_pAVCtx, E_UNEXPECTED);

  int     got_picture = 0;
  int     used_bytes  = 0;
  BOOL    bFlush = (buffer == NULL);
  BOOL    bEndOfSequence = FALSE;

  AVPacket avpkt;
  av_init_packet(&avpkt);

  if (m_pAVCtx->active_thread_type & FF_THREAD_FRAME) {
    if (!m_bFFReordering) {
      m_tcThreadBuffer[m_CurrentThread].rtStart = rtStartIn;
      m_tcThreadBuffer[m_CurrentThread].rtStop  = rtStopIn;
    }

    m_CurrentThread = (m_CurrentThread + 1) % m_pAVCtx->thread_count;
  } else if (m_bBFrameDelay) {
    m_tcBFrameDelay[m_nBFramePos].rtStart = rtStartIn;
    m_tcBFrameDelay[m_nBFramePos].rtStop = rtStopIn;
    m_nBFramePos = !m_nBFramePos;
  }

  uint8_t *pDataBuffer = NULL;
  if (!bFlush && buflen > 0) {
    if (!m_bInputPadded && (!(m_pAVCtx->active_thread_type & FF_THREAD_FRAME) || m_pParser)) {
      // Copy bitstream into temporary buffer to ensure overread protection
      // Verify buffer size
      if (buflen > m_nFFBufferSize) {
        m_nFFBufferSize	= buflen;
        m_pFFBuffer = (BYTE *)av_realloc_f(m_pFFBuffer, m_nFFBufferSize + FF_INPUT_BUFFER_PADDING_SIZE, 1);
        if (!m_pFFBuffer) {
          m_nFFBufferSize = 0;
          return E_OUTOFMEMORY;
        }
      }
      
      memcpy(m_pFFBuffer, buffer, buflen);
      memset(m_pFFBuffer+buflen, 0, FF_INPUT_BUFFER_PADDING_SIZE);
      pDataBuffer = m_pFFBuffer;
    } else {
      pDataBuffer = (uint8_t *)buffer;
    }

    if (m_nCodecId == AV_CODEC_ID_VP8 && m_bWaitingForKeyFrame) {
      if (!(pDataBuffer[0] & 1)) {
        DbgLog((LOG_TRACE, 10, L"::Decode(): Found VP8 key-frame, resuming decoding"));
        m_bWaitingForKeyFrame = FALSE;
      } else {
        return S_OK;
      }
    }
  }

  while (buflen > 0 || bFlush) {
    REFERENCE_TIME rtStart = rtStartIn, rtStop = rtStopIn;

    if (!bFlush) {
      avpkt.data = pDataBuffer;
      avpkt.size = buflen;
      avpkt.pts = rtStartIn;
      if (rtStartIn != AV_NOPTS_VALUE && rtStopIn != AV_NOPTS_VALUE)
        avpkt.duration = (int)(rtStopIn - rtStartIn);
      else
        avpkt.duration = 0;
      avpkt.flags = AV_PKT_FLAG_KEY;

      if (m_bHasPalette) {
        m_bHasPalette = FALSE;
        uint32_t *pal = (uint32_t *)av_packet_new_side_data(&avpkt, AV_PKT_DATA_PALETTE, AVPALETTE_SIZE);
        int pal_size = FFMIN((1 << m_pAVCtx->bits_per_coded_sample) << 2, m_pAVCtx->extradata_size);
        uint8_t *pal_src = m_pAVCtx->extradata + m_pAVCtx->extradata_size - pal_size;

        for (int i = 0; i < pal_size/4; i++)
          pal[i] = 0xFF<<24 | AV_RL32(pal_src+4*i);
      }
    } else {
      avpkt.data = NULL;
      avpkt.size = 0;
    }

    // Parse the data if a parser is present
    // This is mandatory for MPEG-1/2
    if (m_pParser) {
      BYTE *pOut = NULL;
      int pOut_size = 0;

      used_bytes = av_parser_parse2(m_pParser, m_pAVCtx, &pOut, &pOut_size, avpkt.data, avpkt.size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);

      if (used_bytes == 0 && pOut_size == 0 && !bFlush) {
        DbgLog((LOG_TRACE, 50, L"::Decode() - could not process buffer, starving?"));
        break;
      } else if (used_bytes > 0) {
        buflen -= used_bytes;
        pDataBuffer += used_bytes;
      }

      // Update start time cache
//.........这里部分代码省略.........
开发者ID:JERUKA9,项目名称:LAVFilters,代码行数:101,代码来源:avcodec.cpp


示例2: ff_amf_tag_contents

static void ff_amf_tag_contents(void *ctx, const uint8_t *data, const uint8_t *data_end)
{
    int size;
    char buf[1024];

    if (data >= data_end)
        return;
    switch (*data++) {
    case AMF_DATA_TYPE_NUMBER:
        av_log(ctx, AV_LOG_DEBUG, " number %g\n", av_int2double(AV_RB64(data)));
        return;
    case AMF_DATA_TYPE_BOOL:
        av_log(ctx, AV_LOG_DEBUG, " bool %d\n", *data);
        return;
    case AMF_DATA_TYPE_STRING:
    case AMF_DATA_TYPE_LONG_STRING:
        if (data[-1] == AMF_DATA_TYPE_STRING) {
            size = bytestream_get_be16(&data);
        } else {
            size = bytestream_get_be32(&data);
        }
        size = FFMIN(size, 1023);
        memcpy(buf, data, size);
        buf[size] = 0;
        av_log(ctx, AV_LOG_DEBUG, " string '%s'\n", buf);
        return;
    case AMF_DATA_TYPE_NULL:
        av_log(ctx, AV_LOG_DEBUG, " NULL\n");
        return;
    case AMF_DATA_TYPE_ARRAY:
        data += 4;
    case AMF_DATA_TYPE_OBJECT:
        av_log(ctx, AV_LOG_DEBUG, " {\n");
        for (;;) {
            int size = bytestream_get_be16(&data);
            int t;
            memcpy(buf, data, size);
            buf[size] = 0;
            if (!size) {
                av_log(ctx, AV_LOG_DEBUG, " }\n");
                data++;
                break;
            }
            if (data + size >= data_end || data + size < data)
                return;
            data += size;
            av_log(ctx, AV_LOG_DEBUG, "  %s: ", buf);
            ff_amf_tag_contents(ctx, data, data_end);
            t = ff_amf_tag_size(data, data_end);
            if (t < 0 || data + t >= data_end)
                return;
            data += t;
        }
        return;
    case AMF_DATA_TYPE_OBJECT_END:
        av_log(ctx, AV_LOG_DEBUG, " }\n");
        return;
    default:
        return;
    }
}
开发者ID:Fatbag,项目名称:libav,代码行数:61,代码来源:rtmppkt.c


示例3: frame_thread_init

static int frame_thread_init(AVCodecContext *avctx)
{
    int thread_count = avctx->thread_count;
    AVCodec *codec = avctx->codec;
    AVCodecContext *src = avctx;
    FrameThreadContext *fctx;
    int i, err = 0;

    if (!thread_count) {
        int nb_cpus = get_logical_cpus(avctx);
        if ((avctx->debug & (FF_DEBUG_VIS_QP | FF_DEBUG_VIS_MB_TYPE)) || avctx->debug_mv)
            nb_cpus = 1;
        // use number of cores + 1 as thread count if there is more than one
        if (nb_cpus > 1)
            thread_count = avctx->thread_count = FFMIN(nb_cpus + 1, MAX_AUTO_THREADS);
        else
            thread_count = avctx->thread_count = 1;
    }

    if (thread_count <= 1) {
        avctx->active_thread_type = 0;
        return 0;
    }

    avctx->thread_opaque = fctx = av_mallocz(sizeof(FrameThreadContext));

    fctx->threads = av_mallocz(sizeof(PerThreadContext) * thread_count);
    pthread_mutex_init(&fctx->buffer_mutex, NULL);
    fctx->delaying = 1;

    for (i = 0; i < thread_count; i++) {
        AVCodecContext *copy = av_malloc(sizeof(AVCodecContext));
        PerThreadContext *p  = &fctx->threads[i];

        pthread_mutex_init(&p->mutex, NULL);
        pthread_mutex_init(&p->progress_mutex, NULL);
        pthread_cond_init(&p->input_cond, NULL);
        pthread_cond_init(&p->progress_cond, NULL);
        pthread_cond_init(&p->output_cond, NULL);

        p->parent = fctx;
        p->avctx  = copy;

        if (!copy) {
            err = AVERROR(ENOMEM);
            goto error;
        }

        *copy = *src;
        copy->thread_opaque = p;
        copy->pkt = &p->avpkt;

        if (!i) {
            src = copy;

            if (codec->init)
                err = codec->init(copy);

            update_context_from_thread(avctx, copy, 1);
        } else {
            copy->priv_data = av_malloc(codec->priv_data_size);
            if (!copy->priv_data) {
                err = AVERROR(ENOMEM);
                goto error;
            }
            memcpy(copy->priv_data, src->priv_data, codec->priv_data_size);
            copy->internal = av_malloc(sizeof(AVCodecInternal));
            if (!copy->internal) {
                err = AVERROR(ENOMEM);
                goto error;
            }
            *copy->internal = *src->internal;
            copy->internal->is_copy = 1;

            if (codec->init_thread_copy)
                err = codec->init_thread_copy(copy);
        }

        if (err) goto error;

        err = AVERROR(pthread_create(&p->thread, NULL, frame_worker_thread, p));
        p->thread_init= !err;
        if(!p->thread_init)
            goto error;
    }

    return 0;

error:
    frame_thread_free(avctx, i+1);

    return err;
}
开发者ID:Samangan,项目名称:mpc-hc,代码行数:93,代码来源:pthread.c


示例4: multiple_resample

static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
    int i;
    int av_unused mm_flags = av_get_cpu_flags();
    int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 &&
                    (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2;
    int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;

    if (c->compensation_distance)
        dst_size = FFMIN(dst_size, c->compensation_distance);
    src_size = FFMIN(src_size, max_src_size);

    *consumed = 0;

    if (c->filter_length == 1 && c->phase_count == 1) {
        int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index;
        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
        int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;

        dst_size = FFMAX(FFMIN(dst_size, new_size), 0);
        if (dst_size > 0) {
            for (i = 0; i < dst->ch_count; i++) {
                c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);
                if (i+1 == dst->ch_count) {
                    c->index += dst_size * c->dst_incr_div;
                    c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;
                    av_assert2(c->index >= 0);
                    *consumed = c->index;
                    c->frac   = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;
                    c->index = 0;
                }
            }
        }
    } else {
        int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
        int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
        int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
        int (*resample_func)(struct ResampleContext *c, void *dst,
                             const void *src, int n, int update_ctx);

        dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);
        if (dst_size > 0) {
            /* resample_linear and resample_common should have same behavior
             * when frac and dst_incr_mod are zero */
            resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?
                            c->dsp.resample_linear : c->dsp.resample_common;
            for (i = 0; i < dst->ch_count; i++)
                *consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);
        }
    }

    if(need_emms)
        emms_c();

    if (c->compensation_distance) {
        c->compensation_distance -= dst_size;
        if (!c->compensation_distance) {
            c->dst_incr     = c->ideal_dst_incr;
            c->dst_incr_div = c->dst_incr / c->src_incr;
            c->dst_incr_mod = c->dst_incr % c->src_incr;
        }
    }

    return dst_size;
}
开发者ID:jpcottin,项目名称:FFmpeg,代码行数:64,代码来源:resample.c


示例5: decode_frame

static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
                        AVPacket *avpkt) {
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    AVSubtitle *sub = data;
    const uint8_t *buf_end = buf + buf_size;
    uint8_t *bitmap;
    int w, h, x, y, rlelen, i;
    int64_t packet_time = 0;
    GetBitContext gb;

    memset(sub, 0, sizeof(*sub));

    // check that at least header fits
    if (buf_size < 27 + 7 * 2 + 4 * 3) {
        av_log(avctx, AV_LOG_ERROR, "coded frame too small\n");
        return -1;
    }

    // read start and end time
    if (buf[0] != '[' || buf[13] != '-' || buf[26] != ']') {
        av_log(avctx, AV_LOG_ERROR, "invalid time code\n");
        return -1;
    }
    if (avpkt->pts != AV_NOPTS_VALUE)
        packet_time = av_rescale_q(avpkt->pts, AV_TIME_BASE_Q, (AVRational){1, 1000});
    sub->start_display_time = parse_timecode(buf +  1, packet_time);
    sub->end_display_time   = parse_timecode(buf + 14, packet_time);
    buf += 27;

    // read header
    w = bytestream_get_le16(&buf);
    h = bytestream_get_le16(&buf);
    if (avcodec_check_dimensions(avctx, w, h) < 0)
        return -1;
    x = bytestream_get_le16(&buf);
    y = bytestream_get_le16(&buf);

#ifdef SUPPORT_DIVX_DRM
	if((video_height - (y+h)) > 30)
	{
		y = video_height-30-h-1;
	}
#endif /* end of SUPPORT_DIVX_DRM */	
	   
    // skip bottom right position, it gives no new information
    bytestream_get_le16(&buf);
    bytestream_get_le16(&buf);
    rlelen = bytestream_get_le16(&buf);

    // allocate sub and set values
    sub->rects =  av_mallocz(sizeof(*sub->rects));
    sub->rects[0] = av_mallocz(sizeof(*sub->rects[0]));
    sub->num_rects = 1;
    sub->rects[0]->x = x; sub->rects[0]->y = y;
    sub->rects[0]->w = w; sub->rects[0]->h = h;
    sub->rects[0]->type = SUBTITLE_BITMAP;
    sub->rects[0]->pict.linesize[0] = w;
    sub->rects[0]->pict.data[0] = av_malloc(w * h);
    sub->rects[0]->nb_colors = 4;
    sub->rects[0]->pict.data[1] = av_mallocz(AVPALETTE_SIZE);

    // read palette
    for (i = 0; i < sub->rects[0]->nb_colors; i++)
        ((uint32_t*)sub->rects[0]->pict.data[1])[i] = bytestream_get_be24(&buf);
    // make all except background (first entry) non-transparent
#if 1
	if(sub_type == 2)	//DXSA
	{
		for (i = 0; i < sub->rects[0]->nb_colors; i++)
		{
			if(buf[i])
				((uint32_t*)sub->rects[0]->pict.data[1])[i] |= 0xff000000;
		}
		if(buf[0] == buf[1] && buf[0] == buf[2] && buf[0] == buf[3] && buf[1] == buf[2] && buf[1] == buf[3] && buf[2] == buf[3] && buf[0] < 0xff)
		{
			transport_float = (float)buf[0] / 256.0;
		}
		
		buf += 4;
	}
	else
	{
		for (i = 1; i < sub->rects[0]->nb_colors; i++)
			((uint32_t*)sub->rects[0]->pict.data[1])[i] |= 0xff000000;
	}
#else
    for (i = 1; i < sub->rects[0]->nb_colors; i++)
        ((uint32_t*)sub->rects[0]->pict.data[1])[i] |= 0xff000000;
#endif

    // process RLE-compressed data
    rlelen = FFMIN(rlelen, buf_end - buf);
    init_get_bits(&gb, buf, rlelen * 8);
    bitmap = sub->rects[0]->pict.data[0];
    for (y = 0; y < h; y++) {
        // interlaced: do odd lines
        if (y == (h + 1) / 2) bitmap = sub->rects[0]->pict.data[0] + w;
        for (x = 0; x < w; ) {
            int log2 = ff_log2_tab[show_bits(&gb, 8)];
//.........这里部分代码省略.........
开发者ID:dr4g0nsr,项目名称:mplayer-skyviia-8860,代码行数:101,代码来源:xsubdec.c


示例6: qtrle_encode_line

/**
 * Computes the best RLE sequence for a line
 */
static void qtrle_encode_line(QtrleEncContext *s, AVFrame *p, int line, uint8_t **buf)
{
    int width=s->avctx->width;
    int i;
    signed char rlecode;

    /* We will use it to compute the best bulk copy sequence */
    unsigned int bulkcount;
    /* This will be the number of pixels equal to the preivous frame one's
     * starting from the ith pixel */
    unsigned int skipcount;
    /* This will be the number of consecutive equal pixels in the current
     * frame, starting from the ith one also */
    unsigned int repeatcount;

    /* The cost of the three different possibilities */
    int total_bulk_cost;
    int total_skip_cost;
    int total_repeat_cost;

    int temp_cost;
    int j;

    uint8_t *this_line = p->               data[0] + line*p->linesize[0] + (width - 1)*s->pixel_size;
    uint8_t *prev_line = s->previous_frame.data[0] + line*p->linesize[0] + (width - 1)*s->pixel_size;

    s->length_table[width] = 0;
    skipcount = 0;

    for (i = width - 1; i >= 0; i--) {

        if (!s->frame.key_frame && !memcmp(this_line, prev_line, s->pixel_size))
            skipcount = FFMIN(skipcount + 1, MAX_RLE_SKIP);
        else
            skipcount = 0;

        total_skip_cost  = s->length_table[i + skipcount] + 2;
        s->skip_table[i] = skipcount;


        if (i < width - 1 && !memcmp(this_line, this_line + s->pixel_size, s->pixel_size))
            repeatcount = FFMIN(repeatcount + 1, MAX_RLE_REPEAT);
        else
            repeatcount = 1;

        total_repeat_cost = s->length_table[i + repeatcount] + 1 + s->pixel_size;

        /* skip code is free for the first pixel, it costs one byte for repeat and bulk copy
         * so let's make it aware */
        if (i == 0) {
            total_skip_cost--;
            total_repeat_cost++;
        }

        if (repeatcount > 1 && (skipcount == 0 || total_repeat_cost < total_skip_cost)) {
            /* repeat is the best */
            s->length_table[i]  = total_repeat_cost;
            s->rlecode_table[i] = -repeatcount;
        }
        else if (skipcount > 0) {
            /* skip is the best choice here */
            s->length_table[i]  = total_skip_cost;
            s->rlecode_table[i] = 0;
        }
        else {
            /* We cannot do neither skip nor repeat
             * thus we search for the best bulk copy to do */

            int limit = FFMIN(width - i, MAX_RLE_BULK);

            temp_cost = 1 + s->pixel_size + !i;
            total_bulk_cost = INT_MAX;

            for (j = 1; j <= limit; j++) {
                if (s->length_table[i + j] + temp_cost < total_bulk_cost) {
                    /* We have found a better bulk copy ... */
                    total_bulk_cost = s->length_table[i + j] + temp_cost;
                    bulkcount = j;
                }
                temp_cost += s->pixel_size;
            }

            s->length_table[i]  = total_bulk_cost;
            s->rlecode_table[i] = bulkcount;
        }

        this_line -= s->pixel_size;
        prev_line -= s->pixel_size;
    }

    /* Good ! Now we have the best sequence for this line, let's ouput it */

    /* We do a special case for the first pixel so that we avoid testing it in
     * the whole loop */

    i=0;
    this_line = p->               data[0] + line*p->linesize[0];
//.........这里部分代码省略.........
开发者ID:OESF-DLNA,项目名称:upnp-extension,代码行数:101,代码来源:qtrleenc.c


示例7: build_filter


//.........这里部分代码省略.........
        switch(c->format){
        case AV_SAMPLE_FMT_S16P:
            for(i=0;i<tap_count;i++)
                ((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
            if (phase_count % 2) break;
            if (tap_count % 2 == 0 || tap_count == 1) {
                for (i = 0; i < tap_count; i++)
                    ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
            }
            else {
                for (i = 1; i <= tap_count; i++)
                    ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-i] =
                        av_clip_int16(lrintf(tab[i] * scale / (norm - tab[0] + tab[tap_count])));
            }
            break;
        case AV_SAMPLE_FMT_S32P:
            for(i=0;i<tap_count;i++)
                ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
            if (phase_count % 2) break;
            if (tap_count % 2 == 0 || tap_count == 1) {
                for (i = 0; i < tap_count; i++)
                    ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
            }
            else {
                for (i = 1; i <= tap_count; i++)
                    ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-i] =
                        av_clipl_int32(llrint(tab[i] * scale / (norm - tab[0] + tab[tap_count])));
            }
            break;
        case AV_SAMPLE_FMT_FLTP:
            for(i=0;i<tap_count;i++)
                ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
            if (phase_count % 2) break;
            if (tap_count % 2 == 0 || tap_count == 1) {
                for (i = 0; i < tap_count; i++)
                    ((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
            }
            else {
                for (i = 1; i <= tap_count; i++)
                    ((float*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]);
            }
            break;
        case AV_SAMPLE_FMT_DBLP:
            for(i=0;i<tap_count;i++)
                ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
            if (phase_count % 2) break;
            if (tap_count % 2 == 0 || tap_count == 1) {
                for (i = 0; i < tap_count; i++)
                    ((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
            }
            else {
                for (i = 1; i <= tap_count; i++)
                    ((double*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]);
            }
            break;
        }
    }
#if 0
    {
#define LEN 1024
        int j,k;
        double sine[LEN + tap_count];
        double filtered[LEN];
        double maxff=-2, minff=2, maxsf=-2, minsf=2;
        for(i=0; i<LEN; i++){
            double ss=0, sf=0, ff=0;
            for(j=0; j<LEN+tap_count; j++)
                sine[j]= cos(i*j*M_PI/LEN);
            for(j=0; j<LEN; j++){
                double sum=0;
                ph=0;
                for(k=0; k<tap_count; k++)
                    sum += filter[ph * tap_count + k] * sine[k+j];
                filtered[j]= sum / (1<<FILTER_SHIFT);
                ss+= sine[j + center] * sine[j + center];
                ff+= filtered[j] * filtered[j];
                sf+= sine[j + center] * filtered[j];
            }
            ss= sqrt(2*ss/LEN);
            ff= sqrt(2*ff/LEN);
            sf= 2*sf/LEN;
            maxff= FFMAX(maxff, ff);
            minff= FFMIN(minff, ff);
            maxsf= FFMAX(maxsf, sf);
            minsf= FFMIN(minsf, sf);
            if(i%11==0){
                av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
                minff=minsf= 2;
                maxff=maxsf= -2;
            }
        }
    }
#endif

    ret = 0;
fail:
    av_free(tab);
    av_free(sin_lut);
    return ret;
}
开发者ID:jpcottin,项目名称:FFmpeg,代码行数:101,代码来源:resample.c


示例8: uiEvent


//.........这里部分代码省略.........
        break;

    case evForward10sec:
        uiRelSeek(10);
        break;

    case evBackward10sec:
        uiRelSeek(-10);
        break;

    case evSetMoviePosition:
        guiInfo.Position = param;
        uiPctSeek(guiInfo.Position);
        break;

    case evIncVolume:
        mplayer_put_key(KEY_VOLUME_UP);
        break;

    case evDecVolume:
        mplayer_put_key(KEY_VOLUME_DOWN);
        break;

    case evMute:
        mixer_mute(mixer);
        break;

    case evSetVolume:
    case ivSetVolume:
        guiInfo.Volume = param;
        {
            float l = guiInfo.Volume * (100.0 - guiInfo.Balance) / 50.0;
            float r = guiInfo.Volume * guiInfo.Balance / 50.0;
            mixer_setvolume(mixer, FFMIN(l, guiInfo.Volume), FFMIN(r, guiInfo.Volume));
        }

        if (ev == ivSetVolume)
            break;

        if (osd_level) {
            osd_visible = (GetTimerMS() + 1000) | 1;
            vo_osd_progbar_type  = OSD_VOLUME;
            vo_osd_progbar_value = guiInfo.Volume * 256.0 / 100.0;
            vo_osd_changed(OSDTYPE_PROGBAR);
        }

        break;

    case evSetBalance:
    case ivSetBalance:
        guiInfo.Balance = param;
        mixer_setbalance(mixer, (guiInfo.Balance - 50.0) / 50.0);     // transform 0..100 to -1..1
        uiEvent(ivSetVolume, guiInfo.Volume);

        if (ev == ivSetBalance)
            break;

        if (osd_level) {
            osd_visible = (GetTimerMS() + 1000) | 1;
            vo_osd_progbar_type  = OSD_BALANCE;
            vo_osd_progbar_value = guiInfo.Balance * 256.0 / 100.0;
            vo_osd_changed(OSDTYPE_PROGBAR);
        }

        break;
开发者ID:0p1pp1,项目名称:mplayer,代码行数:66,代码来源:actions.c


示例9: while

///returns the number of samples filled in from start.
//also updates data and len to reflect NEW unfilled area - start is unmodified.
int ODFFmpegDecoder::FillDataFromCache(samplePtr & data, sampleFormat outFormat, sampleCount &start, sampleCount& len, unsigned int channel)
{
   if(mDecodeCache.size() <= 0)
      return 0;
   int samplesFilled=0;

   //do a search for the best position to start at.
   //Guess that the array is evenly spaced from end to end - (dictionary sort)
   //assumes the array is sorted.
   //all we need for this to work is a location in the cache array
   //that has a start time of less than our start sample, but try to get closer with binary search
   int searchStart = 0;
   int searchEnd = mDecodeCache.size();
   int guess;
   if(searchEnd>kODFFmpegSearchThreshold)
   {
      //first just guess that the cache is contiguous and we can just use math to figure it out like a dictionary
      //by guessing where our hit will be.
      while(searchStart+1<searchEnd)
      {
         guess = (searchStart+searchEnd)/2;//find a midpoint. //searchStart+ (searchEnd-searchStart)*  ((float)start - mDecodeCache[searchStart]->start )/mDecodeCache[searchEnd]->start;

         //we want guess to point at the first index that hits even if there are duplicate start times (which can happen)
         if(mDecodeCache[guess]->start+mDecodeCache[guess]->len >= start)
            searchEnd = --guess;
         else
            searchStart = guess;
      }
   }

   //this is a sorted array
   for(int i=searchStart; i < (int)mDecodeCache.size(); i++)
   {
      //check for a cache hit - be careful to include the first/last sample an nothing more.
      //we only accept cache hits that touch either end - no piecing out of the middle.
      //this way the amount to be decoded remains set.
      if(start < mDecodeCache[i]->start+mDecodeCache[i]->len &&
         start + len > mDecodeCache[i]->start)
      {
         uint8_t* outBuf;
         outBuf = (uint8_t*)data;
         //reject buffers that would split us into two pieces because we don't have
         //a method of dealing with this yet, and it won't happen very often.
         if(start<mDecodeCache[i]->start && start+len  > mDecodeCache[i]->start+mDecodeCache[i]->len)
            continue;

         int samplesHit;
         int hitStartInCache;
         int hitStartInRequest;
         int nChannels = mDecodeCache[i]->numChannels;
         samplesHit = FFMIN(start+len,mDecodeCache[i]->start+mDecodeCache[i]->len)
                        - FFMAX(mDecodeCache[i]->start,start);
         //find the start of the hit relative to the cache buffer start.
         hitStartInCache   = FFMAX(0,start-mDecodeCache[i]->start);
         //we also need to find out which end was hit - if it is the tail only we need to update from a later index.
         hitStartInRequest = start <mDecodeCache[i]->start?len - samplesHit: 0;
         sampleCount outIndex,inIndex;
         for(int j=0;j<samplesHit;j++)
         {
            outIndex = hitStartInRequest + j;
            inIndex = (hitStartInCache + j) * nChannels + channel;
            switch (mDecodeCache[i]->samplefmt)
            {
               case AV_SAMPLE_FMT_U8:
                  //printf("u8 in %llu out %llu cachelen %llu outLen %llu\n", inIndex, outIndex, mDecodeCache[i]->len, len);
                  ((int16_t *)outBuf)[outIndex] = (int16_t) (((uint8_t*)mDecodeCache[i]->samplePtr)[inIndex] - 0x80) << 8;
               break;

               case AV_SAMPLE_FMT_S16:
                  //printf("u16 in %llu out %llu cachelen %llu outLen %llu\n", inIndex, outIndex, mDecodeCache[i]->len, len);
                  ((int16_t *)outBuf)[outIndex] = ((int16_t*)mDecodeCache[i]->samplePtr)[inIndex];
               break;

               case AV_SAMPLE_FMT_S32:
                  //printf("s32 in %llu out %llu cachelen %llu outLen %llu\n", inIndex, outIndex, mDecodeCache[i]->len, len);
                  ((float *)outBuf)[outIndex] = (float) ((int32_t*)mDecodeCache[i]->samplePtr)[inIndex] * (1.0 / (1 << 31));
               break;

               case AV_SAMPLE_FMT_FLT:
                  //printf("f in %llu out %llu cachelen %llu outLen %llu\n", inIndex, outIndex, mDecodeCache[i]->len, len);
                  ((float *)outBuf)[outIndex] = (float) ((float*)mDecodeCache[i]->samplePtr)[inIndex];
               break;

               case AV_SAMPLE_FMT_DBL:
                  //printf("dbl in %llu out %llu cachelen %llu outLen %llu\n", inIndex, outIndex, mDecodeCache[i]->len, len);
                  ((float *)outBuf)[outIndex] = (float) ((double*)mDecodeCache[i]->samplePtr)[inIndex];
               break;

               default:
                  printf("ODDecodeFFMPEG TASK unrecognized sample format\n");
                  return 1;
               break;
            }
         }
         //update the cursor
         samplesFilled += samplesHit;

         //update the input start/len params - if the end was hit we can take off just len.
//.........这里部分代码省略.........
开发者ID:zhenggzw,项目名称:audacity,代码行数:101,代码来源:ODDecodeFFmpegTask.cpp


示例10: read_header

static int read_header(AVFormatContext *s)
{
    JVDemuxContext *jv = s->priv_data;
    AVIOContext *pb = s->pb;
    AVStream *vst, *ast;
    int64_t audio_pts = 0;
    int64_t offset;
    int i;

    avio_skip(pb, 80);

    ast = avformat_new_stream(s, NULL);
    vst = avformat_new_stream(s, NULL);
    if (!ast || !vst)
        return AVERROR(ENOMEM);

    vst->codec->codec_type  = AVMEDIA_TYPE_VIDEO;
    vst->codec->codec_id    = CODEC_ID_JV;
    vst->codec->codec_tag   = 0; /* no fourcc */
    vst->codec->width       = avio_rl16(pb);
    vst->codec->height      = avio_rl16(pb);
    vst->duration           =
    vst->nb_frames          =
    ast->nb_index_entries   = avio_rl16(pb);
    avpriv_set_pts_info(vst, 64, avio_rl16(pb), 1000);

    avio_skip(pb, 4);

    ast->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
    ast->codec->codec_id    = CODEC_ID_PCM_U8;
    ast->codec->codec_tag   = 0; /* no fourcc */
    ast->codec->sample_rate = avio_rl16(pb);
    ast->codec->channels    = 1;
    avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);

    avio_skip(pb, 10);

    ast->index_entries = av_malloc(ast->nb_index_entries * sizeof(*ast->index_entries));
    if (!ast->index_entries)
        return AVERROR(ENOMEM);

    jv->frames = av_malloc(ast->nb_index_entries * sizeof(JVFrame));
    if (!jv->frames)
        return AVERROR(ENOMEM);

    offset = 0x68 + ast->nb_index_entries * 16;
    for(i = 0; i < ast->nb_index_entries; i++) {
        AVIndexEntry *e   = ast->index_entries + i;
        JVFrame      *jvf = jv->frames + i;

        /* total frame size including audio, video, palette data and padding */
        e->size         = avio_rl32(pb);
        e->timestamp    = i;
        e->pos          = offset;
        offset         += e->size;

        jvf->audio_size = avio_rl32(pb);
        jvf->video_size = avio_rl32(pb);
        jvf->palette_size = avio_r8(pb) ? 768 : 0;
        jvf->video_size = FFMIN(FFMAX(jvf->video_size, 0),
                                INT_MAX - JV_PREAMBLE_SIZE - jvf->palette_size);
        if (avio_r8(pb))
             av_log(s, AV_LOG_WARNING, "unsupported audio codec\n");
        jvf->video_type = avio_r8(pb);
        avio_skip(pb, 1);

        e->timestamp = jvf->audio_size ? audio_pts : AV_NOPTS_VALUE;
        audio_pts += jvf->audio_size;

        e->flags = jvf->video_type != 1 ? AVINDEX_KEYFRAME : 0;
    }

    jv->state = JV_AUDIO;
    return 0;
}
开发者ID:KindDragon,项目名称:FFmpeg,代码行数:75,代码来源:jvdec.c


示例11: main

int main(int argc, char *argv[])
{
    uint64_t i, j;
    uint64_t sse = 0;
    double sse_d = 0.0;
    FILE *f[2];
    uint8_t buf[2][SIZE];
    int len = 1;
    int64_t max;
    int shift      = argc < 5 ? 0 : atoi(argv[4]);
    int skip_bytes = argc < 6 ? 0 : atoi(argv[5]);
    uint64_t size0   = 0;
    uint64_t size1   = 0;
    uint64_t maxdist = 0;
    double maxdist_d = 0.0;

    if (argc < 3) {
        printf("tiny_psnr <file1> <file2> [<elem size> [<shift> [<skip bytes>]]]\n");
        printf("WAV headers are skipped automatically.\n");
        return 1;
    }

    if (argc > 3) {
        if (!strcmp(argv[3], "u8")) {
            len = 1;
        } else if (!strcmp(argv[3], "s16")) {
            len = 2;
        } else if (!strcmp(argv[3], "f32")) {
            len = 4;
        } else if (!strcmp(argv[3], "f64")) {
            len = 8;
        } else {
            char *end;
            len = strtol(argv[3], &end, 0);
            if (*end || len < 1 || len > 2) {
                fprintf(stderr, "Unsupported sample format: %s\n", argv[3]);
                return 1;
            }
        }
    }

    max = (1LL << (8 * len)) - 1;

    f[0] = fopen(argv[1], "rb");
    f[1] = fopen(argv[2], "rb");
    if (!f[0] || !f[1]) {
        fprintf(stderr, "Could not open input files.\n");
        return 1;
    }

    for (i = 0; i < 2; i++) {
        uint8_t *p = buf[i];
        if (fread(p, 1, 12, f[i]) != 12)
            return 1;
        if (!memcmp(p, "RIFF", 4) &&
            !memcmp(p + 8, "WAVE", 4)) {
            if (fread(p, 1, 8, f[i]) != 8)
                return 1;
            while (memcmp(p, "data", 4)) {
                int s = p[4] | p[5] << 8 | p[6] << 16 | p[7] << 24;
                fseek(f[i], s, SEEK_CUR);
                if (fread(p, 1, 8, f[i]) != 8)
                    return 1;
            }
        } else {
            fseek(f[i], -12, SEEK_CUR);
        }
    }

    fseek(f[shift < 0], abs(shift), SEEK_CUR);

    fseek(f[0], skip_bytes, SEEK_CUR);
    fseek(f[1], skip_bytes, SEEK_CUR);

    for (;;) {
        int s0 = fread(buf[0], 1, SIZE, f[0]);
        int s1 = fread(buf[1], 1, SIZE, f[1]);

        for (j = 0; j < FFMIN(s0, s1); j += len) {
            switch (len) {
            case 1:
            case 2: {
                int64_t a = buf[0][j];
                int64_t b = buf[1][j];
                int dist;
                if (len == 2) {
                    a = get_s16l(buf[0] + j);
                    b = get_s16l(buf[1] + j);
                } else {
                    a = buf[0][j];
                    b = buf[1][j];
                }
                sse += (a - b) * (a - b);
                dist = llabs(a - b);
                if (dist > maxdist)
                    maxdist = dist;
                break;
            }
            case 4:
            case 8: {
//.........这里部分代码省略.........
开发者ID:Brainiarc7,项目名称:libav,代码行数:101,代码来源:tiny_psnr.c


示例12: rtp_write_header

static int rtp_write_header(AVFormatContext *s1)
{
    RTPMuxContext *s = s1->priv_data;
    int n, ret = AVERROR(EINVAL);
    AVStream *st;

    if (s1->nb_streams != 1) {
        av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
        return AVERROR(EINVAL);
    }
    st = s1->streams[0];
    if (!is_supported(st->codecpar->codec_id)) {
        av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));

        return -1;
    }

    if (s->payload_type < 0) {
        /* Re-validate non-dynamic payload types */
        if (st->id < RTP_PT_PRIVATE)
            st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);

        s->payload_type = st->id;
    } else {
        /* private option takes priority */
        st->id = s->payload_type;
    }

    s->base_timestamp = av_get_random_seed();
    s->timestamp = s->base_timestamp;
    s->cur_timestamp = 0;
    if (!s->ssrc)
        s->ssrc = av_get_random_seed();
    s->first_packet = 1;
    s->first_rtcp_ntp_time = ff_ntp_time();
    if (s1->start_time_realtime != 0  &&  s1->start_time_realtime != AV_NOPTS_VALUE)
        /* Round the NTP time to whole milliseconds. */
        s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
                                 NTP_OFFSET_US;
    // Pick a random sequence start number, but in the lower end of the
    // available range, so that any wraparound doesn't happen immediately.
    // (Immediate wraparound would be an issue for SRTP.)
    if (s->seq < 0) {
        if (s1->flags & AVFMT_FLAG_BITEXACT) {
            s->seq = 0;
        } else
            s->seq = av_get_random_seed() & 0x0fff;
    } else
        s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval

    if (s1->packet_size) {
        if (s1->pb->max_packet_size)
            s1->packet_size = FFMIN(s1->packet_size,
                                    s1->pb->max_packet_size);
    } else
        s1->packet_size = s1->pb->max_packet_size;
    if (s1->packet_size <= 12) {
        av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
        return AVERROR(EIO);
    }
    s->buf = av_malloc(s1->packet_size);
    if (!s->buf) {
        return AVERROR(ENOMEM);
    }
    s->max_payload_size = s1->packet_size - 12;

    if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
        avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
    } else {
        avpriv_set_pts_info(st, 32, 1, 90000);
    }
    s->buf_ptr = s->buf;
    switch(st->codecpar->codec_id) {
    case AV_CODEC_ID_MP2:
    case AV_CODEC_ID_MP3:
        s->buf_ptr = s->buf + 4;
        avpriv_set_pts_info(st, 32, 1, 90000);
        break;
    case AV_CODEC_ID_MPEG1VIDEO:
    case AV_CODEC_ID_MPEG2VIDEO:
        break;
    case AV_CODEC_ID_MPEG2TS:
        n = s->max_payload_size / TS_PACKET_SIZE;
        if (n < 1)
            n = 1;
        s->max_payload_size = n * TS_PACKET_SIZE;
        break;
    case AV_CODEC_ID_H261:
        if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
            av_log(s, AV_LOG_ERROR,
                   "Packetizing H261 is experimental and produces incorrect "
                   "packetization for cases where GOBs don't fit into packets "
                   "(even though most receivers may handle it just fine). "
                   "Please set -f_strict experimental in order to enable it.\n");
            ret = AVERROR_EXPERIMENTAL;
            goto fail;
        }
        break;
    case AV_CODEC_ID_H264:
        /* check for H.264 MP4 syntax */
//.........这里部分代码省略.........
开发者ID:0day-ci,项目名称:FFmpeg,代码行数:101,代码来源:rtpenc.c


示例13: select_frame

static void select_frame(AVFilterContext *ctx, AVFrame *frame)
{
    SelectContext *select = ctx->priv;
    AVFilterLink *inlink = ctx->inputs[0];
    double res;

    if (isnan(select->var_values[VAR_START_PTS]))
        select->var_values[VAR_START_PTS] = TS2D(frame->pts);
    if (isnan(select->var_values[VAR_START_T]))
        select->var_values[VAR_START_T] = TS2D(frame->pts) * av_q2d(inlink->time_base);

    select->var_values[VAR_N  ] = inlink->frame_count;
    select->var_values[VAR_PTS] = TS2D(frame->pts);
    select->var_values[VAR_T  ] = TS2D(frame->pts) * av_q2d(inlink->time_base);
    select->var_values[VAR_POS] = av_frame_get_pkt_pos(frame) == -1 ? NAN : av_frame_get_pkt_pos(frame);
    select->var_values[VAR_KEY] = frame->key_frame;
    select->var_values[VAR_CONCATDEC_SELECT] = get_concatdec_select(frame, av_rescale_q(frame->pts, inlink->time_base, AV_TIME_BASE_Q));

    switch (inlink->type) {
    case AVMEDIA_TYPE_AUDIO:
        select->var_values[VAR_SAMPLES_N] = frame->nb_samples;
        break;

    case AVMEDIA_TYPE_VIDEO:
        select->var_values[VAR_INTERLACE_TYPE] =
            !frame->interlaced_frame ? INTERLACE_TYPE_P :
        frame->top_field_first ? INTERLACE_TYPE_T : INTERLACE_TYPE_B;
        select->var_values[VAR_PICT_TYPE] = frame->pict_type;
        if (select->do_scene_detect) {
            char buf[32];
            select->var_values[VAR_SCENE] = get_scene_score(ctx, frame);
            // TODO: document metadata
            snprintf(buf, sizeof(buf), "%f", select->var_values[VAR_SCENE]);
            av_dict_set(avpriv_frame_get_metadatap(frame), "lavfi.scene_score", buf, 0);
        }
        break;
    }

    select->select = res = av_expr_eval(select->expr, select->var_values, NULL);
    av_log(inlink->dst, AV_LOG_DEBUG,
           "n:%f pts:%f t:%f key:%d",
           select->var_values[VAR_N],
           select->var_values[VAR_PTS],
           select->var_values[VAR_T],
           frame->key_frame);

    switch (inlink->type) {
    case AVMEDIA_TYPE_VIDEO:
        av_log(inlink->dst, AV_LOG_DEBUG, " interlace_type:%c pict_type:%c scene:%f",
               (!frame->interlaced_frame) ? 'P' :
               frame->top_field_first     ? 'T' : 'B',
               av_get_picture_type_char(frame->pict_type),
               select->var_values[VAR_SCENE]);
        break;
    case AVMEDIA_TYPE_AUDIO:
        av_log(inlink->dst, AV_LOG_DEBUG, " samples_n:%d consumed_samples_n:%f",
               frame->nb_samples,
               select->var_values[VAR_CONSUMED_SAMPLES_N]);
        break;
    }

    if (res == 0) {
        select->select_out = -1; /* drop */
    } else if (isnan(res) || res < 0) {
        select->select_out = 0; /* first output */
    } else {
        select->select_out = FFMIN(ceilf(res)-1, select->nb_outputs-1); /* other outputs */
    }

    av_log(inlink->dst, AV_LOG_DEBUG, " -> select:%f select_out:%d\n", res, select->select_out);

    if (res) {
        select->var_values[VAR_PREV_SELECTED_N]   = select->var_values[VAR_N];
        select->var_values[VAR_PREV_SELECTED_PTS] = select->var_values[VAR_PTS];
        select->var_values[VAR_PREV_SELECTED_T]   = select->var_values[VAR_T];
        select->var_values[VAR_SELECTED_N] += 1.0;
        if (inlink->type == AVMEDIA_TYPE_AUDIO)
            select->var_values[VAR_CONSUMED_SAMPLES_N] += frame->nb_samples;
    }

    select->var_values[VAR_PREV_PTS] = select->var_values[VAR_PTS];
    select->var_values[VAR_PREV_T]   = select->var_values[VAR_T];
}
开发者ID:0day-ci,项目名称:FFmpeg,代码行数:83,代码来源:f_select.c


示例14: flac_parse

static int flac_parse(AVCodecParserContext *s, AVCodecContext *avctx,
                      const uint8_t **poutbuf, int *poutbuf_size,
                      const uint8_t *buf, int buf_size)
{
    FLACParseContext *fpc = s->priv_data;
    FLACHeaderMarker *curr;
    int nb_headers;
    const uint8_t *read_end   = buf;
    const uint8_t *read_start = buf;

    if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
        FLACFrameInfo fi;
        if (frame_header_is_valid(avctx, buf, &fi)) {
            s->duration = fi.blocksize;
            if (!avctx->sample_rate)
                avctx->sample_rate = fi.samplerate;
            if (fpc->pc->flags & PARSER_FLAG_USE_CODEC_TS){
                fpc->pc->pts = fi.frame_or_sample_num;
                if (!fi.is_var_size)
                  fpc->pc->pts *= fi.blocksize;
            }
        }
        *poutbuf      = buf;
        *poutbuf_size = buf_size;
        return buf_size;
    }

    fpc->avctx = avctx;
    if (fpc->best_header_valid)
        return get_best_header(fpc, poutbuf, poutbuf_size);

    /* If a best_header was found last call remove it with the buffer data. */
    if (fpc->best_header && fpc->best_header->best_child) {
        FLACHeaderMarker *temp;
        FLACHeaderMarker *best_child = fpc->best_header->best_child;

        /* Remove headers in list until the end of the best_header. */
        for (curr = fpc->headers; curr != best_child; curr = temp) {
            if (curr != fpc->best_header) {
                av_log(avctx, AV_LOG_DEBUG,
               

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C++ FFT函数代码示例发布时间:2022-05-30
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