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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