本文整理汇总了C++中AV_WL16函数 的典型用法代码示例。如果您正苦于以下问题：C++ AV_WL16函数的具体用法？C++ AV_WL16怎么用？C++ AV_WL16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AV_WL16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: encode_init
static int encode_init(AVCodecContext * avctx){
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2;
uint8_t *extradata;
s->avctx = avctx;
if(avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "too many channels: got %i, need %i or fewer",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if(avctx->bit_rate < 24*1000) {
av_log(avctx, AV_LOG_ERROR, "bitrate too low: got %"PRId64", need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == CODEC_ID_WMAV1) {
extradata= av_malloc(4);
avctx->extradata_size= 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata+2, flags2);
} else if (avctx->codec->id == CODEC_ID_WMAV2) {
extradata= av_mallocz(10);
avctx->extradata_size= 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata+4, flags2);
}else
assert(0);
avctx->extradata= extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
ff_wma_init(avctx, flags2);
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
avctx->block_align=
s->block_align= avctx->bit_rate*s->frame_len / (avctx->sample_rate*8);
//av_log(NULL, AV_LOG_ERROR, "%d %"PRId64" %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
avctx->frame_size= s->frame_len;
return 0;
}开发者ID:eugenehp， 项目名称:ffmbc， 代码行数:52， 代码来源:wmaenc.c
示例2: gamma_convert
static void gamma_convert(uint8_t * src[], int width, uint16_t *gamma)
{
int i;
uint16_t *src1 = (uint16_t*)src[0];
for (i = 0; i < width; ++i) {
uint16_t r = AV_RL16(src1 + i*4 + 0);
uint16_t g = AV_RL16(src1 + i*4 + 1);
uint16_t b = AV_RL16(src1 + i*4 + 2);
AV_WL16(src1 + i*4 + 0, gamma[r]);
AV_WL16(src1 + i*4 + 1, gamma[g]);
AV_WL16(src1 + i*4 + 2, gamma[b]);
}
}开发者ID:TeamHint， 项目名称:FFmpeg， 代码行数:15， 代码来源:swscale.c
示例3: encode_init
static int encode_init(AVCodecContext * avctx){
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2;
uint8_t *extradata;
s->avctx = avctx;
if(avctx->channels > MAX_CHANNELS)
return -1;
if(avctx->bit_rate < 24*1000)
return -1;
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == CODEC_ID_WMAV1) {
extradata= av_malloc(4);
avctx->extradata_size= 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata+2, flags2);
} else if (avctx->codec->id == CODEC_ID_WMAV2) {
extradata= av_mallocz(10);
avctx->extradata_size= 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata+4, flags2);
}else
assert(0);
avctx->extradata= extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
ff_wma_init(avctx, flags2);
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0);
avctx->block_align=
s->block_align= avctx->bit_rate*(int64_t)s->frame_len / (avctx->sample_rate*8);
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
avctx->frame_size= s->frame_len;
return 0;
}开发者ID:OESF-DLNA， 项目名称:upnp-extension， 代码行数:46， 代码来源:wmaenc.c
示例4: wsaud_read_packet
static int wsaud_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
AVIOContext *pb = s->pb;
unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
unsigned int chunk_size;
int ret = 0;
AVStream *st = s->streams[0];
if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
AUD_CHUNK_PREAMBLE_SIZE)
return AVERROR(EIO);
/* validate the chunk */
if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
return AVERROR_INVALIDDATA;
chunk_size = AV_RL16(&preamble[0]);
if (st->codec->codec_id == CODEC_ID_WESTWOOD_SND1) {
/* For Westwood SND1 audio we need to add the output size and input
size to the start of the packet to match what is in VQA.
Specifically, this is needed to signal when a packet should be
decoding as raw 8-bit pcm or variable-size ADPCM. */
int out_size = AV_RL16(&preamble[2]);
if ((ret = av_new_packet(pkt, chunk_size + 4)))
return ret;
if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
return ret < 0 ? ret : AVERROR(EIO);
AV_WL16(&pkt->data[0], out_size);
AV_WL16(&pkt->data[2], chunk_size);
pkt->duration = out_size;
} else {
ret = av_get_packet(pb, pkt, chunk_size);
if (ret != chunk_size)
return AVERROR(EIO);
/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
pkt->duration = (chunk_size * 2) / st->codec->channels;
}
pkt->stream_index = st->index;
return ret;
}开发者ID:stainberg， 项目名称:android_FFMPEG， 代码行数:45， 代码来源:westwood_aud.c
示例5: siff_read_packet
static int siff_read_packet(AVFormatContext *s, AVPacket *pkt)
{
SIFFContext *c = s->priv_data;
int size;
if (c->has_video)
{
if (c->cur_frame >= c->frames)
return AVERROR(EIO);
if (c->curstrm == -1)
{
c->pktsize = avio_rl32(s->pb) - 4;
c->flags = avio_rl16(s->pb);
c->gmcsize = (c->flags & VB_HAS_GMC) ? 4 : 0;
if (c->gmcsize)
avio_read(s->pb, c->gmc, c->gmcsize);
c->sndsize = (c->flags & VB_HAS_AUDIO) ? avio_rl32(s->pb): 0;
c->curstrm = !!(c->flags & VB_HAS_AUDIO);
}
if (!c->curstrm)
{
size = c->pktsize - c->sndsize;
if (av_new_packet(pkt, size) < 0)
return AVERROR(ENOMEM);
AV_WL16(pkt->data, c->flags);
if (c->gmcsize)
memcpy(pkt->data + 2, c->gmc, c->gmcsize);
avio_read(s->pb, pkt->data + 2 + c->gmcsize, size - c->gmcsize - 2);
pkt->stream_index = 0;
c->curstrm = -1;
}
else
{
if (av_get_packet(s->pb, pkt, c->sndsize - 4) < 0)
return AVERROR(EIO);
pkt->stream_index = 1;
c->curstrm = 0;
}
if(!c->cur_frame || c->curstrm)
pkt->flags |= AV_PKT_FLAG_KEY;
if (c->curstrm == -1)
c->cur_frame++;
}
else
{
size = av_get_packet(s->pb, pkt, c->block_align);
if(size <= 0)
return AVERROR(EIO);
}
return pkt->size;
}开发者ID:pengdu， 项目名称:freescale_omx_framework， 代码行数:52， 代码来源:siff.c
示例6: rgbpack_fields
static void rgbpack_fields(void *ctx_,
uint8_t *src[AVS_MAX_COMPONENTS],
int sstrides[AVS_MAX_COMPONENTS],
uint8_t *dst[AVS_MAX_COMPONENTS],
int dstrides[AVS_MAX_COMPONENTS],
int w, int h)
{
RGBPackContext *ctx = ctx_;
uint8_t *rgb[3], *dest;
unsigned val;
int i, j, c;
rgb[0] = src[0];
rgb[1] = src[1];
rgb[2] = src[2];
dest = dst[0];
for (j = 0; j < h; j++) {
for (i = 0; i < w; i++) {
val = 0;
if (ctx->inbpp <= 8) {
for (c = 0; c < 3; c++)
val |= rgb[c][i] << ctx->shift[c];
} else {
for (c = 0; c < 3; c++)
val |= AV_RN16(rgb[c] + i * 2) << ctx->shift[c];
}
switch (ctx->step) {
case 1:
dest[i] = val;
break;
case 2:
if (ctx->be) AV_WB16(dest + i * 2, val);
else AV_WL16(dest + i * 2, val);
break;
case 4:
if (ctx->be) AV_WB32(dest + i * 4, val);
else AV_WL32(dest + i * 4, val);
break;
}
}
for (c = 0; c < 3; c++)
rgb[c] += sstrides[0];
dest += dstrides[0];
}
}开发者ID:lu-zero， 项目名称:avscale， 代码行数:46， 代码来源:rgbpck.c
示例7: dfa_read_header
static int dfa_read_header(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
AVStream *st;
int frames;
int version;
uint32_t mspf;
if (avio_rl32(pb) != MKTAG('D', 'F', 'I', 'A')) {
av_log(s, AV_LOG_ERROR, "Invalid magic for DFA\n");
return AVERROR_INVALIDDATA;
}
version = avio_rl16(pb);
frames = avio_rl16(pb);
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
st->codec->codec_id = AV_CODEC_ID_DFA;
st->codec->width = avio_rl16(pb);
st->codec->height = avio_rl16(pb);
mspf = avio_rl32(pb);
if (!mspf) {
av_log(s, AV_LOG_WARNING, "Zero FPS reported, defaulting to 10\n");
mspf = 100;
}
avpriv_set_pts_info(st, 24, mspf, 1000);
avio_skip(pb, 128 - 16); // padding
st->duration = frames;
if (ff_alloc_extradata(st->codec, 2))
return AVERROR(ENOMEM);
AV_WL16(st->codec->extradata, version);
if (version == 0x100)
st->sample_aspect_ratio = (AVRational){2, 1};
return 0;
}开发者ID:Bjelijah， 项目名称:EcamTurnH265， 代码行数:41， 代码来源:dfa.c
示例8: encode_init
static int encode_init(AVCodecContext * avctx){
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2;
uint8_t *extradata;
s->avctx = avctx;
if(avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "too many channels: got %i, need %i or fewer",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if (avctx->sample_rate > 48000) {
av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if(avctx->bit_rate < 24*1000) {
av_log(avctx, AV_LOG_ERROR, "bitrate too low: got %i, need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
extradata= av_malloc(4);
avctx->extradata_size= 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata+2, flags2);
} else if (avctx->codec->id == AV_CODEC_ID_WMAV2) {
extradata= av_mallocz(10);
avctx->extradata_size= 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata+4, flags2);
}else
assert(0);
avctx->extradata= extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
if (avctx->channels == 2)
s->ms_stereo = 1;
ff_wma_init(avctx, flags2);
/* init MDCT */
for(i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
s->block_align = avctx->bit_rate * (int64_t)s->frame_len /
(avctx->sample_rate * 8);
s->block_align = FFMIN(s->block_align, MAX_CODED_SUPERFRAME_SIZE);
avctx->block_align = s->block_align;
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len;
avctx->frame_size = avctx->delay = s->frame_len;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = &s->frame;
avcodec_get_frame_defaults(avctx->coded_frame);
#endif
return 0;
}开发者ID:JSinglan， 项目名称:libav， 代码行数:68， 代码来源:wmaenc.c
示例9: bind_lavc
static int bind_lavc(audio_encoder_t *encoder, muxer_stream_t *mux_a)
{
mux_a->wf = malloc(sizeof(WAVEFORMATEX)+lavc_actx->extradata_size+256);
mux_a->wf->wFormatTag = lavc_param_atag;
mux_a->wf->nChannels = lavc_actx->channels;
mux_a->wf->nSamplesPerSec = lavc_actx->sample_rate;
mux_a->wf->nAvgBytesPerSec = (lavc_actx->bit_rate / 8);
mux_a->avg_rate= lavc_actx->bit_rate;
mux_a->h.dwRate = mux_a->wf->nAvgBytesPerSec;
if(lavc_actx->block_align)
mux_a->h.dwSampleSize = mux_a->h.dwScale = lavc_actx->block_align;
else
{
mux_a->h.dwScale = (mux_a->wf->nAvgBytesPerSec * lavc_actx->frame_size)/ mux_a->wf->nSamplesPerSec; /* for cbr */
if ((mux_a->wf->nAvgBytesPerSec *
lavc_actx->frame_size) % mux_a->wf->nSamplesPerSec)
{
mux_a->h.dwScale = lavc_actx->frame_size;
mux_a->h.dwRate = lavc_actx->sample_rate;
mux_a->h.dwSampleSize = 0; // Blocksize not constant
}
else
mux_a->h.dwSampleSize = 0;
}
if(mux_a->h.dwSampleSize)
mux_a->wf->nBlockAlign = mux_a->h.dwSampleSize;
else
mux_a->wf->nBlockAlign = 1;
mux_a->h.dwSuggestedBufferSize = (encoder->params.audio_preload*mux_a->wf->nAvgBytesPerSec)/1000;
mux_a->h.dwSuggestedBufferSize -= mux_a->h.dwSuggestedBufferSize % mux_a->wf->nBlockAlign;
switch(lavc_param_atag)
{
case 0x11: /* imaadpcm */
mux_a->wf->wBitsPerSample = 4;
mux_a->wf->cbSize = 2;
AV_WL16(mux_a->wf+1, lavc_actx->frame_size);
break;
case 0x55: /* mp3 */
mux_a->wf->cbSize = 12;
mux_a->wf->wBitsPerSample = 0; /* does not apply */
((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->wID = 1;
((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->fdwFlags = 2;
((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nBlockSize = mux_a->wf->nBlockAlign;
((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nFramesPerBlock = 1;
((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nCodecDelay = 0;
break;
default:
mux_a->wf->wBitsPerSample = 0; /* Unknown */
if (lavc_actx->extradata && (lavc_actx->extradata_size > 0))
{
memcpy(mux_a->wf+1, lavc_actx->extradata, lavc_actx->extradata_size);
mux_a->wf->cbSize = lavc_actx->extradata_size;
}
else
mux_a->wf->cbSize = 0;
break;
}
// Fix allocation
mux_a->wf = realloc(mux_a->wf, sizeof(WAVEFORMATEX)+mux_a->wf->cbSize);
encoder->input_format = AF_FORMAT_S16_NE;
encoder->min_buffer_size = mux_a->h.dwSuggestedBufferSize;
encoder->max_buffer_size = mux_a->h.dwSuggestedBufferSize*2;
return 1;
}开发者ID:azuwis， 项目名称:mplayer， 代码行数:69， 代码来源:ae_lavc.c
示例10: smush_read_header
//.........这里部分代码省略.........
nframes = avio_rl32(pb);
if (!nframes)
return AVERROR_INVALIDDATA;
avio_skip(pb, 2); // skip pad
width = avio_rl16(pb);
height = avio_rl16(pb);
avio_skip(pb, 2); // skip pad
avio_skip(pb, size - 14);
if (avio_rb32(pb) != MKBETAG('F', 'L', 'H', 'D'))
return AVERROR_INVALIDDATA;
size = avio_rb32(pb);
while (!got_audio && ((read + 8) < size)) {
uint32_t sig, chunk_size;
if (avio_feof(pb))
return AVERROR_EOF;
sig = avio_rb32(pb);
chunk_size = avio_rb32(pb);
read += 8;
switch (sig) {
case MKBETAG('W', 'a', 'v', 'e'):
got_audio = 1;
sample_rate = avio_rl32(pb);
if (!sample_rate)
return AVERROR_INVALIDDATA;
channels = avio_rl32(pb);
if (!channels)
return AVERROR_INVALIDDATA;
avio_skip(pb, chunk_size - 8);
read += chunk_size;
break;
case MKBETAG('B', 'l', '1', '6'):
case MKBETAG('A', 'N', 'N', 'O'):
avio_skip(pb, chunk_size);
read += chunk_size;
break;
default:
return AVERROR_INVALIDDATA;
break;
}
}
avio_skip(pb, size - read);
} else {
av_log(ctx, AV_LOG_ERROR, "Wrong magic\n");
return AVERROR_INVALIDDATA;
}
vst = avformat_new_stream(ctx, 0);
if (!vst)
return AVERROR(ENOMEM);
smush->video_stream_index = vst->index;
avpriv_set_pts_info(vst, 64, 1, 15);
vst->start_time = 0;
vst->duration =
vst->nb_frames = nframes;
vst->avg_frame_rate = av_inv_q(vst->time_base);
vst->codec->codec_type = AVMEDIA_TYPE_VIDEO;
vst->codec->codec_id = AV_CODEC_ID_SANM;
vst->codec->codec_tag = 0;
vst->codec->width = width;
vst->codec->height = height;
if (!smush->version) {
if (ff_alloc_extradata(vst->codec, 1024 + 2))
return AVERROR(ENOMEM);
AV_WL16(vst->codec->extradata, subversion);
for (i = 0; i < 256; i++)
AV_WL32(vst->codec->extradata + 2 + i * 4, palette[i]);
}
if (got_audio) {
ast = avformat_new_stream(ctx, 0);
if (!ast)
return AVERROR(ENOMEM);
smush->audio_stream_index = ast->index;
ast->start_time = 0;
ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
ast->codec->codec_id = AV_CODEC_ID_ADPCM_VIMA;
ast->codec->codec_tag = 0;
ast->codec->sample_rate = sample_rate;
ast->codec->channels = channels;
avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
}
return 0;
}开发者ID:twinaphex， 项目名称:vice-libretro， 代码行数:101， 代码来源:smush.c
示例11: encode_init
static av_cold int encode_init(AVCodecContext *avctx)
{
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2, block_align;
uint8_t *extradata;
int ret;
s->avctx = avctx;
if (avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR,
"too many channels: got %i, need %i or fewer\n",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if (avctx->sample_rate > 48000) {
av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz\n",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if (avctx->bit_rate < 24 * 1000) {
av_log(avctx, AV_LOG_ERROR,
"bitrate too low: got %i, need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
extradata = av_malloc(4);
if (!extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata + 2, flags2);
} else if (avctx->codec->id == AV_CODEC_ID_WMAV2) {
extradata = av_mallocz(10);
if (!extradata)
return AVERROR(ENOMEM);
avctx->extradata_size = 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata + 4, flags2);
} else {
av_assert0(0);
}
avctx->extradata = extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
if (avctx->channels == 2)
s->ms_stereo = 1;
if ((ret = ff_wma_init(avctx, flags2)) < 0)
return ret;
/* init MDCT */
for (i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
block_align = avctx->bit_rate * (int64_t) s->frame_len /
(avctx->sample_rate * 8);
block_align = FFMIN(block_align, MAX_CODED_SUPERFRAME_SIZE);
avctx->block_align = block_align;
avctx->frame_size = avctx->initial_padding = s->frame_len;
return 0;
}开发者ID:artclarke， 项目名称:humble-video， 代码行数:71， 代码来源:wmaenc.c
示例12: decode_frame
//.........这里部分代码省略.........
}
if (s->stripsizesoff) {
if (s->stripsizesoff >= (unsigned)avpkt->size)
return AVERROR_INVALIDDATA;
bytestream2_init(&stripsizes, avpkt->data + s->stripsizesoff,
avpkt->size - s->stripsizesoff);
}
if (s->strippos) {
if (s->strippos >= (unsigned)avpkt->size)
return AVERROR_INVALIDDATA;
bytestream2_init(&stripdata, avpkt->data + s->strippos,
avpkt->size - s->strippos);
}
if (s->rps <= 0) {
av_log(avctx, AV_LOG_ERROR, "rps %d invalid\n", s->rps);
return AVERROR_INVALIDDATA;
}
planes = s->planar ? s->bppcount : 1;
for (plane = 0; plane < planes; plane++) {
stride = p->linesize[plane];
dst = p->data[plane];
for (i = 0; i < s->height; i += s->rps) {
if (s->stripsizesoff)
ssize = ff_tget(&stripsizes, s->sstype, le);
else
ssize = s->stripsize;
if (s->strippos)
soff = ff_tget(&stripdata, s->sot, le);
else
soff = s->stripoff;
if (soff > avpkt->size || ssize > avpkt->size - soff) {
av_log(avctx, AV_LOG_ERROR, "Invalid strip size/offset\n");
return AVERROR_INVALIDDATA;
}
if ((ret = tiff_unpack_strip(s, p, dst, stride, avpkt->data + soff, ssize, i,
FFMIN(s->rps, s->height - i))) < 0) {
if (avctx->err_recognition & AV_EF_EXPLODE)
return ret;
break;
}
dst += s->rps * stride;
}
if (s->predictor == 2) {
if (s->photometric == TIFF_PHOTOMETRIC_YCBCR) {
av_log(s->avctx, AV_LOG_ERROR, "predictor == 2 with YUV is unsupported");
return AVERROR_PATCHWELCOME;
}
dst = p->data[plane];
soff = s->bpp >> 3;
if (s->planar)
soff = FFMAX(soff / s->bppcount, 1);
ssize = s->width * soff;
if (s->avctx->pix_fmt == AV_PIX_FMT_RGB48LE ||
s->avctx->pix_fmt == AV_PIX_FMT_RGBA64LE ||
s->avctx->pix_fmt == AV_PIX_FMT_GBRP16LE ||
s->avctx->pix_fmt == AV_PIX_FMT_GBRAP16LE) {
for (i = 0; i < s->height; i++) {
for (j = soff; j < ssize; j += 2)
AV_WL16(dst + j, AV_RL16(dst + j) + AV_RL16(dst + j - soff));
dst += stride;
}
} else if (s->avctx->pix_fmt == AV_PIX_FMT_RGB48BE ||
s->avctx->pix_fmt == AV_PIX_FMT_RGBA64BE ||
s->avctx->pix_fmt == AV_PIX_FMT_GBRP16BE ||
s->avctx->pix_fmt == AV_PIX_FMT_GBRAP16BE) {
for (i = 0; i < s->height; i++) {
for (j = soff; j < ssize; j += 2)
AV_WB16(dst + j, AV_RB16(dst + j) + AV_RB16(dst + j - soff));
dst += stride;
}
} else {
for (i = 0; i < s->height; i++) {
for (j = soff; j < ssize; j++)
dst[j] += dst[j - soff];
dst += stride;
}
}
}
if (s->photometric == TIFF_PHOTOMETRIC_WHITE_IS_ZERO) {
dst = p->data[plane];
for (i = 0; i < s->height; i++) {
for (j = 0; j < p->linesize[plane]; j++)
dst[j] = (s->avctx->pix_fmt == AV_PIX_FMT_PAL8 ? (1<<s->bpp) - 1 : 255) - dst[j];
dst += stride;
}
}
}
if (s->planar && s->bppcount > 2) {
FFSWAP(uint8_t*, p->data[0], p->data[2]);
FFSWAP(int, p->linesize[0], p->linesize[2]);
FFSWAP(uint8_t*, p->data[0], p->data[1]);
FFSWAP(int, p->linesize[0], p->linesize[1]);
}开发者ID:venkatarajasekhar， 项目名称:Qt， 代码行数:101， 代码来源:tiff.c
示例13: filter_out
static int filter_out(struct af_instance *af)
{
af_ac3enc_t *s = af->priv;
if (!s->pending)
return 0;
AVFrame *frame = av_frame_alloc();
if (!frame) {
MP_FATAL(af, "Could not allocate memory \n");
return -1;
}
int err = -1;
AVPacket pkt = {0};
av_init_packet(&pkt);
#if HAVE_AVCODEC_NEW_CODEC_API
// Send input as long as it wants.
while (1) {
err = read_input_frame(af, frame);
if (err < 0)
goto done;
if (err == 0)
break;
err = -1;
int lavc_ret = avcodec_send_frame(s->lavc_actx, frame);
// On EAGAIN, we're supposed to read remaining output.
if (lavc_ret == AVERROR(EAGAIN))
break;
if (lavc_ret < 0) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
s->encoder_buffered += s->input->samples;
s->input->samples = 0;
}
int lavc_ret = avcodec_receive_packet(s->lavc_actx, &pkt);
if (lavc_ret == AVERROR(EAGAIN)) {
// Need to buffer more input.
err = 0;
goto done;
}
if (lavc_ret < 0) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
#else
err = read_input_frame(af, frame);
if (err < 0)
goto done;
if (err == 0)
goto done;
err = -1;
int ok;
int lavc_ret = avcodec_encode_audio2(s->lavc_actx, &pkt, frame, &ok);
s->input->samples = 0;
if (lavc_ret < 0 || !ok) {
MP_FATAL(af, "Encode failed.\n");
goto done;
}
#endif
MP_DBG(af, "avcodec_encode_audio got %d, pending %d.\n",
pkt.size, s->pending->samples + s->input->samples);
s->encoder_buffered -= AC3_FRAME_SIZE;
struct mp_audio *out =
mp_audio_pool_get(af->out_pool, af->data, s->out_samples);
if (!out)
goto done;
mp_audio_copy_attributes(out, s->pending);
int frame_size = pkt.size;
int header_len = 0;
char hdr[8];
if (s->cfg_add_iec61937_header && pkt.size > 5) {
int bsmod = pkt.data[5] & 0x7;
int len = frame_size;
frame_size = AC3_FRAME_SIZE * 2 * 2;
header_len = 8;
AV_WL16(hdr, 0xF872); // iec 61937 syncword 1
AV_WL16(hdr + 2, 0x4E1F); // iec 61937 syncword 2
hdr[5] = bsmod; // bsmod
hdr[4] = 0x01; // data-type ac3
AV_WL16(hdr + 6, len << 3); // number of bits in payload
}
if (frame_size > out->samples * out->sstride)
abort();
char *buf = (char *)out->planes[0];
memcpy(buf, hdr, header_len);
memcpy(buf + header_len, pkt.data, pkt.size);
memset(buf + header_len + pkt.size, 0,
frame_size - (header_len + pkt.size));
//.........这里部分代码省略.........
开发者ID:chyiz， 项目名称:mpv， 代码行数:101， 代码来源:af_lavcac3enc.c
示例14: wsvqa_read_packet
static int wsvqa_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
WsVqaDemuxContext *wsvqa = s->priv_data;
AVIOContext *pb = s->pb;
int ret = -1;
unsigned char preamble[VQA_PREAMBLE_SIZE];
unsigned int chunk_type;
unsigned int chunk_size;
int skip_byte;
while (avio_read(pb, preamble, VQA_PREAMBLE_SIZE) == VQA_PREAMBLE_SIZE) {
chunk_type = AV_RB32(&preamble[0]);
chunk_size = AV_RB32(&preamble[4]);
skip_byte = chunk_size & 0x01;
if ((chunk_type == SND0_TAG) || (chunk_type == SND1_TAG) ||
(chunk_type == SND2_TAG) || (chunk_type == VQFR_TAG)) {
if (av_new_packet(pkt, chunk_size))
return AVERROR(EIO);
ret = avio_read(pb, pkt->data, chunk_size);
if (ret != chunk_size) {
av_free_packet(pkt);
return AVERROR(EIO);
}
switch (chunk_type) {
case SND0_TAG:
case SND1_TAG:
case SND2_TAG:
if (wsvqa->audio_stream_index == -1) {
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
wsvqa->audio_stream_index = st->index;
if (!wsvqa->sample_rate)
wsvqa->sample_rate = 22050;
if (!wsvqa->channels)
wsvqa->channels = 1;
if (!wsvqa->bps)
wsvqa->bps = 8;
st->codec->sample_rate = wsvqa->sample_rate;
st->codec->bits_per_coded_sample = wsvqa->bps;
st->codec->channels = wsvqa->channels;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
switch (chunk_type) {
case SND0_TAG:
if (wsvqa->bps == 16)
st->codec->codec_id = AV_CODEC_ID_PCM_S16LE;
else
st->codec->codec_id = AV_CODEC_ID_PCM_U8;
break;
case SND1_TAG:
st->codec->codec_id = AV_CODEC_ID_WESTWOOD_SND1;
break;
case SND2_TAG:
st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_WS;
st->codec->extradata_size = 2;
st->codec->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE);
if (!st->codec->extradata)
return AVERROR(ENOMEM);
AV_WL16(st->codec->extradata, wsvqa->version);
break;
}
}
pkt->stream_index = wsvqa->audio_stream_index;
switch (chunk_type) {
case SND1_TAG:
/* unpacked size is stored in header */
pkt->duration = AV_RL16(pkt->data) / wsvqa->channels;
break;
case SND2_TAG:
/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
pkt->duration = (chunk_size * 2) / wsvqa->channels;
break;
}
break;
case VQFR_TAG:
pkt->stream_index = wsvqa->video_stream_index;
pkt->duration = 1;
break;
}
/* stay on 16-bit alignment */
if (skip_byte)
avio_skip(pb, 1);
return ret;
} else {
switch(chunk_type){
case CMDS_TAG:
break;
default:
av_log(s, AV_LOG_INFO, "Skipping unknown chunk 0x%08X\n", chunk_type);
//.........这里部分代码省略.........
开发者ID:AVbin， 项目名称:libav， 代码行数:101， 代码来源:westwood_vqa.c
示例15: filter_slice
static int filter_slice(RotContext *rot, ThreadData *td, int job, int nb_jobs)
{
AVFrame *in = td->in;
AVFrame *out = td->out;
const int outw = td->outw, outh = td->outh;
const int inw = td->inw, inh = td->inh;
const int plane = td->plane;
const int xi = td->xi, yi = td->yi;
const int c = td->c, s = td->s;
const int start = (outh * job ) / nb_jobs;
const int end = (outh * (job+1)) / nb_jobs;
int xprime = td->xprime + start * s;
int yprime = td->yprime + start * c;
int i, j, x, y;
for (j = start; j < end; j++) {
x = xprime + xi + FIXP*(inw-1)/2;
y = yprime + yi + FIXP*(inh-1)/2;
if (fabs(rot->angle - 0) < FLT_EPSILON && outw == inw && outh == inh) {
simple_rotate(out->data[plane] + j * out->linesize[plane],
in->data[plane] + j * in->linesize[plane],
in->linesize[plane], 0, rot->draw.pixelstep[plane], outw);
} else if (fabs(rot->angle - M_PI/2) < FLT_EPSILON && outw == inh && outh == inw) {
simple_rotate(out->data[plane] + j * out->linesize[plane],
in->data[plane] + j * rot->draw.pixelstep[plane],
in->linesize[plane], 1, rot->draw.pixelstep[plane], outw);
} else if (fabs(rot->angle - M_PI) < FLT_EPSILON && outw == inw && outh == inh) {
simple_rotate(out->data[plane] + j * out->linesize[plane],
in->data[plane] + (outh-j-1) * in->linesize[plane],
in->linesize[plane], 2, rot->draw.pixelstep[plane], outw);
} else if (fabs(rot->angle - 3*M_PI/2) < FLT_EPSILON && outw == inh && outh == inw) {
simple_rotate(out->data[plane] + j * out->linesize[plane],
in->data[plane] + (outh-j-1) * rot->draw.pixelstep[plane],
in->linesize[plane], 3, rot->draw.pixelstep[plane], outw);
} else {
for (i = 0; i < outw; i++) {
int32_t v;
int x1, y1;
uint8_t *pin, *pout;
x1 = x>>16;
y1 = y>>16;
/* the out-of-range values avoid border artifacts */
if (x1 >= -1 && x1 <= inw && y1 >= -1 && y1 <= inh) {
uint8_t inp_inv[4]; /* interpolated input value */
pout = out->data[plane] + j * out->linesize[plane] + i * rot->draw.pixelstep[plane];
if (rot->use_bilinear) {
pin = rot->interpolate_bilinear(inp_inv,
in->data[plane], in->linesize[plane], rot->draw.pixelstep[plane],
x, y, inw-1, inh-1);
} else {
int x2 = av_clip(x1, 0, inw-1);
int y2 = av_clip(y1, 0, inh-1);
pin = in->data[plane] + y2 * in->linesize[plane] + x2 * rot->draw.pixelstep[plane];
}
switch (rot->draw.pixelstep[plane]) {
case 1:
*pout = *pin;
break;
case 2:
v = AV_RL16(pin);
AV_WL16(pout, v);
break;
case 3:
v = AV_RB24(pin);
AV_WB24(pout, v);
break;
case 4:
*((uint32_t *)pout) = *((uint32_t *)pin);
break;
default:
memcpy(pout, pin, rot->draw.pixelstep[plane]);
break;
}
}
x += c;
y -= s;
}
}
xprime += s;
yprime += c;
}
return 0;
}开发者ID:ginozh， 项目名称:my_wmm， 代码行数:87， 代码来源:filter_rotate.cpp
示例16: targa_encode_frame
static int targa_encode_frame(AVCodecContext *avctx,
unsigned char *outbuf,
int buf_size, void *data){
AVFrame *p = (AVFrame *)data;
int bpp, picsize, datasize = -1;
uint8_t *out;
if(avctx->width > 0xffff || avctx->height > 0xffff) {
av_log(avctx, AV_LOG_ERROR, "image dimensions too large\n");
return AVERROR(EINVAL);
}
picsize = avpicture_get_size(avctx->pix_fmt, avctx->width, avctx->height);
if(buf_size < picsize + 45) {
av_log(avctx, AV_LOG_ERROR, "encoded frame too large\n");
return AVERROR(EINVAL);
}
p->pict_type= AV_PICTURE_TYPE_I;
p->key_frame= 1;
/* zero out the header and only set applicable fields */
memset(outbuf, 0, 12);
AV_WL16(outbuf+12, avctx->width);
AV_WL16(outbuf+14, avctx->height);
/* image descriptor byte: origin is always top-left, bits 0-3 specify alpha */
outbuf[17] = 0x20 | (avctx->pix_fmt == PIX_FMT_BGRA ? 8 : 0);
switch(avctx->pix_fmt) {
case PIX_FMT_GRAY8:
outbuf[2] = TGA_BW; /* uncompressed grayscale image */
outbuf[16] = 8; /* bpp */
break;
case PIX_FMT_RGB555LE:
outbuf[2] = TGA_RGB; /* uncompresses true-color image */
outbuf[16] = 16; /* bpp */
break;
case PIX_FMT_BGR24:
outbuf[2] = TGA_RGB; /* uncompressed true-color image */
outbuf[16] = 24; /* bpp */
break;
case PIX_FMT_BGRA:
outbuf[2] = TGA_RGB; /* uncompressed true-color image */
outbuf[16] = 32; /* bpp */
break;
default:
av_log(avctx, AV_LOG_ERROR, "Pixel format '%s' not supported.\n",
av_get_pix_fmt_name(avctx->pix_fmt));
return AVERROR(EINVAL);
}
bpp = outbuf[16] >> 3;
out = outbuf + 18; /* skip past the header we just output */
/* try RLE compression */
if (avctx->coder_type != FF_CODER_TYPE_RAW)
datasize = targa_encode_rle(out, picsize, p, bpp, avctx->width, avctx->height);
/* if that worked well, mark the picture as RLE compressed */
if(datasize >= 0)
outbuf[2] |= 8;
/* if RLE didn't make it smaller, go back to no compression */
else datasize = targa_encode_normal(out, p, bpp, avctx->width, avctx->height);
out += datasize;
/* The standard recommends including this section, even if we don't use
* any of the features it affords. TODO: take advantage of the pixel
* aspect ratio and encoder ID fields available? */
memcpy(out, "\0\0\0\0\0\0\0\0TRUEVISION-XFILE.", 26);
return out + 26 - outbuf;
}开发者ID:mcodegeeks， 项目名称:OpenKODE-Framework， 代码行数:73， 代码来源:targaenc.cpp
示例17: peak_write_frame
static void peak_write_frame(AVFormatContext *s)
{
WAVMuxContext *wav = s->priv_data;
AVCodecContext *enc = s->streams[0]->codec;
int peak_of_peaks;
int c;
if (!wav->peak_output)
return;
for (c = 0; c < enc->channels; c++) {
wav->peak_maxneg[c] = -wav->peak_maxneg[c];
if (wav->peak_bps == 2 && wav->peak_format == PEAK_FORMAT_UINT8) {
wav->peak_maxpos[c] = wav->peak_maxpos[c] / 256;
wav->peak_maxneg[c] = wav->peak_maxneg[c] / 256;
}
if (wav->peak_ppv == 1)
wav->peak_maxpos[c] =
FFMAX(wav->peak_maxpos[c], wav->peak_maxneg[c]);
peak_of_peaks = FFMAX3(wav->peak_maxpos[c], wav->peak_maxneg[c],
wav->peak_pop);
if (peak_of_peaks > wav->peak_pop)
wav->peak_pos_pop = wav->peak_num_frames;
wav->peak_pop = peak_of_peaks;
if (wav->peak_outbuf_size - wav->peak_outbuf_bytes <
wav->peak_format * wav->peak_ppv) {
wav->peak_outbuf_size += PEAK_BUFFER_SIZE;
wav->peak_output = av_realloc(wav->peak_output,
wav->peak_outbuf_size);
if (!wav->peak_output) {
av_log(s, AV_LOG_ERROR, "No memory for peak data\n");
return;
}
}
if (wav->peak_format == PEAK_FORMAT_UINT8) {
wav->peak_output[wav->peak_outbuf_bytes++] =
wav->peak_maxpos[c];
if (wav->peak_ppv == 2) {
wav->peak_output[wav->peak_outbuf_bytes++] =
wav->peak_maxneg[c];
}
} else {
AV_WL16(wav->peak_output + wav->peak_outbuf_bytes,
wav->peak_maxpos[c]);
wav->peak_outbuf_bytes += 2;
if (wav->peak_ppv == 2) {
AV_WL16(wav->peak_output + wav->peak_outbuf_bytes,
wav->peak_maxneg[c]);
wav->peak_outbuf_bytes += 2;
}
}
wav->peak_maxpos[c] = 0;
wav->peak_maxneg[c] = 0;
}
wav->peak_num_frames++;
}开发者ID:26mansi， 项目名称:FFmpeg， 代码行数:61， 代码来源:wavenc.c
示例18: load_ipmovie_packet
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