这篇教程C++ AV_RL32函数代码示例写得很实用，希望能帮到您。
本文整理汇总了C++中AV_RL32函数的典型用法代码示例。如果您正苦于以下问题：C++ AV_RL32函数的具体用法？C++ AV_RL32怎么用？C++ AV_RL32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_RL32函数的30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: SkipFilestatic int SkipFile(struct stream *s, int *count, rar_file_t ***file, const rar_block_t *hdr, const char *volume_mrl){ int min_size = 7+21; if (hdr->flags & RAR_BLOCK_FILE_HAS_HIGH) min_size += 8; if (hdr->size < (unsigned)min_size) return -1; bstr data = stream_peek(s, min_size); if (data.len < min_size) return -1; const uint8_t *peek = (uint8_t *)data.start; /* */ uint32_t file_size_low = AV_RL32(&peek[7+4]); uint8_t method = peek[7+18]; uint16_t name_size = AV_RL16(&peek[7+19]); uint32_t file_size_high = 0; if (hdr->flags & RAR_BLOCK_FILE_HAS_HIGH) file_size_high = AV_RL32(&peek[7+29]); const uint64_t file_size = ((uint64_t)file_size_high << 32) | file_size_low; char *name = calloc(1, name_size + 1); if (!name) return -1; const int name_offset = (hdr->flags & RAR_BLOCK_FILE_HAS_HIGH) ? (7+33) : (7+25); if (name_offset + name_size <= hdr->size) { const int max_size = name_offset + name_size; bstr namedata = stream_peek(s, max_size); if (namedata.len < max_size) { free(name); return -1; } memcpy(name, &namedata.start[name_offset], name_size); } rar_file_t *current = NULL; if (method != 0x30) { MP_WARN(s, "Ignoring compressed file %s (method=0x%2.2x)/n", name, method); goto exit; } /* */ if( *count > 0 ) current = (*file)[*count - 1]; if (current && (current->is_complete || strcmp(current->name, name) || (hdr->flags & RAR_BLOCK_FILE_HAS_PREVIOUS) == 0)) current = NULL; if (!current) { if (hdr->flags & RAR_BLOCK_FILE_HAS_PREVIOUS) goto exit; current = calloc(1, sizeof(*current)); if (!current) goto exit; MP_TARRAY_APPEND(NULL, *file, *count, current); current->name = name; current->size = file_size; current->is_complete = false; current->real_size = 0; current->chunk_count = 0; current->chunk = NULL; name = NULL; } /* Append chunks */ rar_file_chunk_t *chunk = malloc(sizeof(*chunk)); if (chunk) { chunk->mrl = strdup(volume_mrl); chunk->offset = stream_tell(s) + hdr->size; chunk->size = hdr->add_size; chunk->cummulated_size = 0; if (current->chunk_count > 0) { rar_file_chunk_t *previous = current->chunk[current->chunk_count-1]; chunk->cummulated_size += previous->cummulated_size + previous->size; } MP_TARRAY_APPEND(NULL, current->chunk, current->chunk_count, chunk); current->real_size += hdr->add_size; } if ((hdr->flags & RAR_BLOCK_FILE_HAS_NEXT) == 0) current->is_complete = true;exit: /* */ free(name); /* We stop on the first non empty file if we cannot seek */ if (current) { bool can_seek = s->end_pos > 0;//.........这里部分代码省略.........
开发者ID:lu-zero，项目名称:mpv，代码行数:101，
示例2: str_read_packetstatic int str_read_packet(AVFormatContext *s, AVPacket *ret_pkt){ AVIOContext *pb = s->pb; StrDemuxContext *str = s->priv_data; unsigned char sector[RAW_CD_SECTOR_SIZE]; int channel; AVPacket *pkt; AVStream *st; while (1) { if (avio_read(pb, sector, RAW_CD_SECTOR_SIZE) != RAW_CD_SECTOR_SIZE) return AVERROR(EIO); channel = sector[0x11]; if (channel >= 32) return AVERROR_INVALIDDATA; switch (sector[0x12] & CDXA_TYPE_MASK) { case CDXA_TYPE_DATA: case CDXA_TYPE_VIDEO: { int current_sector = AV_RL16(§or[0x1C]); int sector_count = AV_RL16(§or[0x1E]); int frame_size = AV_RL32(§or[0x24]); if(!( frame_size>=0 && current_sector < sector_count && sector_count*VIDEO_DATA_CHUNK_SIZE >=frame_size)){ av_log(s, AV_LOG_ERROR, "Invalid parameters %d %d %d/n", current_sector, sector_count, frame_size); break; } if(str->channels[channel].video_stream_index < 0){ /* allocate a new AVStream */ st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); avpriv_set_pts_info(st, 64, 1, 15); str->channels[channel].video_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->codec->codec_id = CODEC_ID_MDEC; st->codec->codec_tag = 0; /* no fourcc */ st->codec->width = AV_RL16(§or[0x28]); st->codec->height = AV_RL16(§or[0x2A]); } /* if this is the first sector of the frame, allocate a pkt */ pkt = &str->channels[channel].tmp_pkt; if(pkt->size != sector_count*VIDEO_DATA_CHUNK_SIZE){ if(pkt->data) av_log(s, AV_LOG_ERROR, "missmatching sector_count/n"); av_free_packet(pkt); if (av_new_packet(pkt, sector_count*VIDEO_DATA_CHUNK_SIZE)) return AVERROR(EIO); pkt->pos= avio_tell(pb) - RAW_CD_SECTOR_SIZE; pkt->stream_index = str->channels[channel].video_stream_index; } memcpy(pkt->data + current_sector*VIDEO_DATA_CHUNK_SIZE, sector + VIDEO_DATA_HEADER_SIZE, VIDEO_DATA_CHUNK_SIZE); if (current_sector == sector_count-1) { pkt->size= frame_size; *ret_pkt = *pkt; pkt->data= NULL; pkt->size= -1; return 0; } } break; case CDXA_TYPE_AUDIO: if(str->channels[channel].audio_stream_index < 0){ int fmt = sector[0x13]; /* allocate a new AVStream */ st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); str->channels[channel].audio_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_ADPCM_XA; st->codec->codec_tag = 0; /* no fourcc */ st->codec->channels = (fmt&1)?2:1; st->codec->sample_rate = (fmt&4)?18900:37800; // st->codec->bit_rate = 0; //FIXME; st->codec->block_align = 128;//.........这里部分代码省略.........
开发者ID:KindDragon，项目名称:FFmpeg，代码行数:101，
示例3: decode_rlestatic int decode_rle(CamtasiaContext *c, unsigned int srcsize){ unsigned char *src = c->decomp_buf; unsigned char *output, *output_end; int p1, p2, line=c->height, pos=0, i; uint16_t pix16; uint32_t pix32; output = c->pic.data[0] + (c->height - 1) * c->pic.linesize[0]; output_end = c->pic.data[0] + (c->height) * c->pic.linesize[0]; while(src < c->decomp_buf + srcsize) { p1 = *src++; if(p1 == 0) { //Escape code p2 = *src++; if(p2 == 0) { //End-of-line output = c->pic.data[0] + (--line) * c->pic.linesize[0]; if (line < 0) return -1; pos = 0; continue; } else if(p2 == 1) { //End-of-picture return 0; } else if(p2 == 2) { //Skip p1 = *src++; p2 = *src++; line -= p2; if (line < 0) return -1; pos += p1; output = c->pic.data[0] + line * c->pic.linesize[0] + pos * (c->bpp / 8); continue; } // Copy data if (output + p2 * (c->bpp / 8) > output_end) { src += p2 * (c->bpp / 8); continue; } if ((c->bpp == 8) || (c->bpp == 24)) { for(i = 0; i < p2 * (c->bpp / 8); i++) { *output++ = *src++; } // RLE8 copy is actually padded - and runs are not! if(c->bpp == 8 && (p2 & 1)) { src++; } } else if (c->bpp == 16) { for(i = 0; i < p2; i++) { pix16 = AV_RL16(src); src += 2; *(uint16_t*)output = pix16; output += 2; } } else if (c->bpp == 32) { for(i = 0; i < p2; i++) { pix32 = AV_RL32(src); src += 4; *(uint32_t*)output = pix32; output += 4; } } pos += p2; } else { //Run of pixels int pix[4]; //original pixel switch(c->bpp){ case 8: pix[0] = *src++; break; case 16: pix16 = AV_RL16(src); src += 2; *(uint16_t*)pix = pix16; break; case 24: pix[0] = *src++; pix[1] = *src++; pix[2] = *src++; break; case 32: pix32 = AV_RL32(src); src += 4; *(uint32_t*)pix = pix32; break; } if (output + p1 * (c->bpp / 8) > output_end) continue; for(i = 0; i < p1; i++) { switch(c->bpp){ case 8: *output++ = pix[0]; break; case 16: *(uint16_t*)output = pix16; output += 2; break; case 24: *output++ = pix[0]; *output++ = pix[1]; *output++ = pix[2]; break; case 32: *(uint32_t*)output = pix32; output += 4; break; } } pos += p1; } }//.........这里部分代码省略.........
开发者ID:BOTCrusher，项目名称:sagetv，代码行数:101，
示例4: flic_decode_frame_15_16BPPstatic int flic_decode_frame_15_16BPP(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf, int buf_size){ /* Note, the only difference between the 15Bpp and 16Bpp */ /* Format is the pixel format, the packets are processed the same. */ FlicDecodeContext *s = avctx->priv_data; int stream_ptr = 0; int pixel_ptr; unsigned char palette_idx1; unsigned int frame_size; int num_chunks; unsigned int chunk_size; int chunk_type; int i, j; int lines; int compressed_lines; signed short line_packets; int y_ptr; int byte_run; int pixel_skip; int pixel_countdown; unsigned char *pixels; int pixel; unsigned int pixel_limit; s->frame.reference = 1; s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE; if (avctx->reget_buffer(avctx, &s->frame) < 0) { av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed/n"); return -1; } pixels = s->frame.data[0]; pixel_limit = s->avctx->height * s->frame.linesize[0]; frame_size = AV_RL32(&buf[stream_ptr]); stream_ptr += 6; /* skip the magic number */ num_chunks = AV_RL16(&buf[stream_ptr]); stream_ptr += 10; /* skip padding */ frame_size -= 16; /* iterate through the chunks */ while ((frame_size > 0) && (num_chunks > 0)) { chunk_size = AV_RL32(&buf[stream_ptr]); stream_ptr += 4; chunk_type = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; switch (chunk_type) { case FLI_256_COLOR: case FLI_COLOR: /* For some reason, it seems that non-palettized flics do * include one of these chunks in their first frame. * Why I do not know, it seems rather extraneous. *//* av_log(avctx, AV_LOG_ERROR, "Unexpected Palette chunk %d in non-paletised FLC/n",chunk_type);*/ stream_ptr = stream_ptr + chunk_size - 6; break; case FLI_DELTA: case FLI_DTA_LC: y_ptr = 0; compressed_lines = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; while (compressed_lines > 0) { line_packets = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; if (line_packets < 0) { line_packets = -line_packets; y_ptr += line_packets * s->frame.linesize[0]; } else { compressed_lines--; pixel_ptr = y_ptr; CHECK_PIXEL_PTR(0); pixel_countdown = s->avctx->width; for (i = 0; i < line_packets; i++) { /* account for the skip bytes */ pixel_skip = buf[stream_ptr++]; pixel_ptr += (pixel_skip*2); /* Pixel is 2 bytes wide */ pixel_countdown -= pixel_skip; byte_run = (signed char)(buf[stream_ptr++]); if (byte_run < 0) { byte_run = -byte_run; pixel = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; CHECK_PIXEL_PTR(2 * byte_run); for (j = 0; j < byte_run; j++, pixel_countdown -= 2) { *((signed short*)(&pixels[pixel_ptr])) = pixel; pixel_ptr += 2; } } else { CHECK_PIXEL_PTR(2 * byte_run); for (j = 0; j < byte_run; j++, pixel_countdown--) { *((signed short*)(&pixels[pixel_ptr])) = AV_RL16(&buf[stream_ptr]);//.........这里部分代码省略.........
开发者ID:Unhelpful，项目名称:ffmpeg，代码行数:101，
示例5: smka_decode_frame/** * Decode Smacker audio data */static int smka_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt){ SmackerAudioContext *s = avctx->priv_data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; GetBitContext gb; HuffContext h[4]; VLC vlc[4]; int16_t *samples; uint8_t *samples8; int val; int i, res, ret; int unp_size; int bits, stereo; int pred[2] = {0, 0}; if (buf_size <= 4) { av_log(avctx, AV_LOG_ERROR, "packet is too small/n"); return AVERROR(EINVAL); } unp_size = AV_RL32(buf); init_get_bits(&gb, buf + 4, (buf_size - 4) * 8); if(!get_bits1(&gb)){ av_log(avctx, AV_LOG_INFO, "Sound: no data/n"); *got_frame_ptr = 0; return 1; } stereo = get_bits1(&gb); bits = get_bits1(&gb); if (stereo ^ (avctx->channels != 1)) { av_log(avctx, AV_LOG_ERROR, "channels mismatch/n"); return AVERROR(EINVAL); } if (bits && avctx->sample_fmt == AV_SAMPLE_FMT_U8) { av_log(avctx, AV_LOG_ERROR, "sample format mismatch/n"); return AVERROR(EINVAL); } /* get output buffer */ s->frame.nb_samples = unp_size / (avctx->channels * (bits + 1)); if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed/n"); return ret; } samples = (int16_t *)s->frame.data[0]; samples8 = s->frame.data[0]; memset(vlc, 0, sizeof(VLC) * 4); memset(h, 0, sizeof(HuffContext) * 4); // Initialize for(i = 0; i < (1 << (bits + stereo)); i++) { h[i].length = 256; h[i].maxlength = 0; h[i].current = 0; h[i].bits = av_mallocz(256 * 4); h[i].lengths = av_mallocz(256 * sizeof(int)); h[i].values = av_mallocz(256 * sizeof(int)); skip_bits1(&gb); smacker_decode_tree(&gb, &h[i], 0, 0); skip_bits1(&gb); if(h[i].current > 1) { res = init_vlc(&vlc[i], SMKTREE_BITS, h[i].length, h[i].lengths, sizeof(int), sizeof(int), h[i].bits, sizeof(uint32_t), sizeof(uint32_t), INIT_VLC_LE); if(res < 0) { av_log(avctx, AV_LOG_ERROR, "Cannot build VLC table/n"); return -1; } } } if(bits) { //decode 16-bit data for(i = stereo; i >= 0; i--) pred[i] = av_bswap16(get_bits(&gb, 16)); for(i = 0; i <= stereo; i++) *samples++ = pred[i]; for(; i < unp_size / 2; i++) { if(get_bits_left(&gb)<0) return -1; if(i & stereo) { if(vlc[2].table) res = get_vlc2(&gb, vlc[2].table, SMKTREE_BITS, 3); else res = 0; val = h[2].values[res]; if(vlc[3].table) res = get_vlc2(&gb, vlc[3].table, SMKTREE_BITS, 3); else res = 0; val |= h[3].values[res] << 8; pred[1] += sign_extend(val, 16); *samples++ = av_clip_int16(pred[1]); } else { if(vlc[0].table)//.........这里部分代码省略.........
开发者ID:1ibraheem，项目名称:xbmc，代码行数:101，
示例6: decode_framestatic int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt){ const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; DPXContext *const s = avctx->priv_data; AVFrame *picture = data; AVFrame *const p = &s->picture; uint8_t *ptr; int magic_num, offset, endian; int x, y; int w, h, stride, bits_per_color, descriptor, elements, target_packet_size, source_packet_size; unsigned int rgbBuffer; magic_num = AV_RB32(buf); buf += 4; /* Check if the files "magic number" is "SDPX" which means it uses * big-endian or XPDS which is for little-endian files */ if (magic_num == AV_RL32("SDPX")) { endian = 0; } else if (magic_num == AV_RB32("SDPX")) { endian = 1; } else { av_log(avctx, AV_LOG_ERROR, "DPX marker not found/n"); return -1; } offset = read32(&buf, endian); // Need to end in 0x304 offset from start of file buf = avpkt->data + 0x304; w = read32(&buf, endian); h = read32(&buf, endian); // Need to end in 0x320 to read the descriptor buf += 20; descriptor = buf[0]; // Need to end in 0x323 to read the bits per color buf += 3; avctx->bits_per_raw_sample = bits_per_color = buf[0]; switch (descriptor) { case 51: // RGBA elements = 4; break; case 50: // RGB elements = 3; break; default: av_log(avctx, AV_LOG_ERROR, "Unsupported descriptor %d/n", descriptor); return -1; } switch (bits_per_color) { case 8: if (elements == 4) { avctx->pix_fmt = PIX_FMT_RGBA; } else { avctx->pix_fmt = PIX_FMT_RGB24; } source_packet_size = elements; target_packet_size = elements; break; case 10: avctx->pix_fmt = PIX_FMT_RGB48; target_packet_size = 6; source_packet_size = elements * 2; break; case 12: case 16: if (endian) { avctx->pix_fmt = PIX_FMT_RGB48BE; } else { avctx->pix_fmt = PIX_FMT_RGB48LE; } target_packet_size = 6; source_packet_size = elements * 2; break; default: av_log(avctx, AV_LOG_ERROR, "Unsupported color depth : %d/n", bits_per_color); return -1; } if (s->picture.data[0]) avctx->release_buffer(avctx, &s->picture); if (avcodec_check_dimensions(avctx, w, h)) return -1; if (w != avctx->width || h != avctx->height) avcodec_set_dimensions(avctx, w, h); if (avctx->get_buffer(avctx, p) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed/n"); return -1; }//.........这里部分代码省略.........
开发者ID:WangCrystal，项目名称:FFplayer，代码行数:101，
示例7: fourxm_read_headerstatic int fourxm_read_header(AVFormatContext *s){ AVIOContext *pb = s->pb; unsigned int fourcc_tag; unsigned int size; int header_size; FourxmDemuxContext *fourxm = s->priv_data; unsigned char *header; int i, ret; AVStream *st; fourxm->track_count = 0; fourxm->tracks = NULL; fourxm->fps = 1.0; /* skip the first 3 32-bit numbers */ avio_skip(pb, 12); /* check for LIST-HEAD */ GET_LIST_HEADER(); header_size = size - 4; if (fourcc_tag != HEAD_TAG || header_size < 0) return AVERROR_INVALIDDATA; /* allocate space for the header and load the whole thing */ header = av_malloc(header_size); if (!header) return AVERROR(ENOMEM); if (avio_read(pb, header, header_size) != header_size){ av_free(header); return AVERROR(EIO); } /* take the lazy approach and search for any and all vtrk and strk chunks */ for (i = 0; i < header_size - 8; i++) { fourcc_tag = AV_RL32(&header[i]); size = AV_RL32(&header[i + 4]); if (size > header_size - i - 8 && (fourcc_tag == vtrk_TAG || fourcc_tag == strk_TAG)) { av_log(s, AV_LOG_ERROR, "chunk larger than array %d>%d/n", size, header_size - i - 8); return AVERROR_INVALIDDATA; } if (fourcc_tag == std__TAG) { fourxm->fps = av_int2float(AV_RL32(&header[i + 12])); } else if (fourcc_tag == vtrk_TAG) { /* check that there is enough data */ if (size != vtrk_SIZE) { ret= AVERROR_INVALIDDATA; goto fail; } fourxm->width = AV_RL32(&header[i + 36]); fourxm->height = AV_RL32(&header[i + 40]); /* allocate a new AVStream */ st = avformat_new_stream(s, NULL); if (!st){ ret= AVERROR(ENOMEM); goto fail; } avpriv_set_pts_info(st, 60, 1, fourxm->fps); fourxm->video_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->codec->codec_id = AV_CODEC_ID_4XM; st->codec->extradata_size = 4; st->codec->extradata = av_malloc(4); AV_WL32(st->codec->extradata, AV_RL32(&header[i + 16])); st->codec->width = fourxm->width; st->codec->height = fourxm->height; i += 8 + size; } else if (fourcc_tag == strk_TAG) { int current_track; /* check that there is enough data */ if (size != strk_SIZE) { ret= AVERROR_INVALIDDATA; goto fail; } current_track = AV_RL32(&header[i + 8]); if((unsigned)current_track >= UINT_MAX / sizeof(AudioTrack) - 1){ av_log(s, AV_LOG_ERROR, "current_track too large/n"); ret = AVERROR_INVALIDDATA; goto fail; } if (current_track + 1 > fourxm->track_count) { fourxm->tracks = av_realloc_f(fourxm->tracks, sizeof(AudioTrack), current_track + 1); if (!fourxm->tracks) { ret = AVERROR(ENOMEM); goto fail; } memset(&fourxm->tracks[fourxm->track_count], 0, sizeof(AudioTrack) * (current_track + 1 - fourxm->track_count)); fourxm->track_count = current_track + 1; } fourxm->tracks[current_track].adpcm = AV_RL32(&header[i + 12]); fourxm->tracks[current_track].channels = AV_RL32(&header[i + 36]); fourxm->tracks[current_track].sample_rate = AV_RL32(&header[i + 40]);//.........这里部分代码省略.........
开发者ID:Brhett，项目名称:FFmpeg，代码行数:101，
示例8: flic_read_packetstatic int flic_read_packet(AVFormatContext *s, AVPacket *pkt){ FlicDemuxContext *flic = s->priv_data; ByteIOContext *pb = s->pb; int packet_read = 0; unsigned int size; int magic; int ret = 0; unsigned char preamble[FLIC_PREAMBLE_SIZE]; while (!packet_read) { if ((ret = get_buffer(pb, preamble, FLIC_PREAMBLE_SIZE)) != FLIC_PREAMBLE_SIZE) { ret = AVERROR(EIO); break; } size = AV_RL32(&preamble[0]); magic = AV_RL16(&preamble[4]); if (((magic == FLIC_CHUNK_MAGIC_1) || (magic == FLIC_CHUNK_MAGIC_2)) && size > FLIC_PREAMBLE_SIZE) { if (av_new_packet(pkt, size)) { ret = AVERROR(EIO); break; } pkt->stream_index = flic->video_stream_index; pkt->pts = flic->frame_number++; pkt->pos = url_ftell(pb); memcpy(pkt->data, preamble, FLIC_PREAMBLE_SIZE); ret = get_buffer(pb, pkt->data + FLIC_PREAMBLE_SIZE, size - FLIC_PREAMBLE_SIZE); if (ret != size - FLIC_PREAMBLE_SIZE) { av_free_packet(pkt); ret = AVERROR(EIO); } packet_read = 1; } else if (magic == FLIC_TFTD_CHUNK_AUDIO) { if (av_new_packet(pkt, size)) { ret = AVERROR(EIO); break; } /* skip useless 10B sub-header (yes, it's not accounted for in the chunk header) */ url_fseek(pb, 10, SEEK_CUR); pkt->stream_index = flic->audio_stream_index; pkt->pos = url_ftell(pb); ret = get_buffer(pb, pkt->data, size); if (ret != size) { av_free_packet(pkt); ret = AVERROR(EIO); } packet_read = 1; } else { /* not interested in this chunk */ url_fseek(pb, size - 6, SEEK_CUR); } } return ret;}
开发者ID:119，项目名称:dropcam_for_iphone，代码行数:65，
示例9: flic_read_headerstatic int flic_read_header(AVFormatContext *s, AVFormatParameters *ap){ FlicDemuxContext *flic = s->priv_data; ByteIOContext *pb = s->pb; unsigned char header[FLIC_HEADER_SIZE]; AVStream *st, *ast; int speed; int magic_number; unsigned char preamble[FLIC_PREAMBLE_SIZE]; flic->frame_number = 0; /* load the whole header and pull out the width and height */ if (get_buffer(pb, header, FLIC_HEADER_SIZE) != FLIC_HEADER_SIZE) return AVERROR(EIO); magic_number = AV_RL16(&header[4]); speed = AV_RL32(&header[0x10]); if (speed == 0) speed = FLIC_DEFAULT_SPEED; /* initialize the decoder streams */ st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); flic->video_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->codec->codec_id = CODEC_ID_FLIC; st->codec->codec_tag = 0; /* no fourcc */ st->codec->width = AV_RL16(&header[0x08]); st->codec->height = AV_RL16(&header[0x0A]); if (!st->codec->width || !st->codec->height) { /* Ugly hack needed for the following sample: */ /* http://samples.mplayerhq.hu/fli-flc/fli-bugs/specular.flc */ av_log(s, AV_LOG_WARNING, "File with no specified width/height. Trying 640x480./n"); st->codec->width = 640; st->codec->height = 480; } /* send over the whole 128-byte FLIC header */ st->codec->extradata_size = FLIC_HEADER_SIZE; st->codec->extradata = av_malloc(FLIC_HEADER_SIZE); memcpy(st->codec->extradata, header, FLIC_HEADER_SIZE); /* peek at the preamble to detect TFTD videos - they seem to always start with an audio chunk */ if (get_buffer(pb, preamble, FLIC_PREAMBLE_SIZE) != FLIC_PREAMBLE_SIZE) { av_log(s, AV_LOG_ERROR, "Failed to peek at preamble/n"); return AVERROR(EIO); } url_fseek(pb, -FLIC_PREAMBLE_SIZE, SEEK_CUR); /* Time to figure out the framerate: * If the first preamble's magic number is 0xAAAA then this file is from * X-COM: Terror from the Deep. If on the other hand there is a FLIC chunk * magic number at offset 0x10 assume this file is from Magic Carpet instead. * If neither of the above is true then this is a normal FLIC file. */ if (AV_RL16(&preamble[4]) == FLIC_TFTD_CHUNK_AUDIO) { /* TFTD videos have an extra 22050 Hz 8-bit mono audio stream */ ast = av_new_stream(s, 1); if (!ast) return AVERROR(ENOMEM); flic->audio_stream_index = ast->index; /* all audio frames are the same size, so use the size of the first chunk for block_align */ ast->codec->block_align = AV_RL32(&preamble[0]); ast->codec->codec_type = CODEC_TYPE_AUDIO; ast->codec->codec_id = CODEC_ID_PCM_U8; ast->codec->codec_tag = 0; ast->codec->sample_rate = FLIC_TFTD_SAMPLE_RATE; ast->codec->channels = 1; ast->codec->sample_fmt = SAMPLE_FMT_U8; ast->codec->bit_rate = st->codec->sample_rate * 8; ast->codec->bits_per_coded_sample = 8; ast->codec->channel_layout = CH_LAYOUT_MONO; ast->codec->extradata_size = 0; /* Since the header information is incorrect we have to figure out the * framerate using block_align and the fact that the audio is 22050 Hz. * We usually have two cases: 2205 -> 10 fps and 1470 -> 15 fps */ av_set_pts_info(st, 64, ast->codec->block_align, FLIC_TFTD_SAMPLE_RATE); av_set_pts_info(ast, 64, 1, FLIC_TFTD_SAMPLE_RATE); } else if (AV_RL16(&header[0x10]) == FLIC_CHUNK_MAGIC_1) { av_set_pts_info(st, 64, FLIC_MC_SPEED, 70); /* rewind the stream since the first chunk is at offset 12 */ url_fseek(pb, 12, SEEK_SET); /* send over abbreviated FLIC header chunk */ av_free(st->codec->extradata); st->codec->extradata_size = 12; st->codec->extradata = av_malloc(12); memcpy(st->codec->extradata, header, 12); } else if (magic_number == FLIC_FILE_MAGIC_1) {//.........这里部分代码省略.........
开发者ID:119，项目名称:dropcam_for_iphone，代码行数:101，
示例10: get_tcp_server_response/** Read incoming MMST media, header or command packet. */static MMSSCPacketType get_tcp_server_response(MMSContext *mms){ int read_result; MMSSCPacketType packet_type= -1; for(;;) { read_result = url_read_complete(mms->mms_hd, mms->in_buffer, 8); if (read_result != 8) { if(read_result < 0) { av_log(NULL, AV_LOG_ERROR, "Error reading packet header: %d (%s)/n", read_result, strerror(read_result)); packet_type = SC_PKT_CANCEL; } else { av_log(NULL, AV_LOG_ERROR, "The server closed the connection/n"); packet_type = SC_PKT_NO_DATA; } return packet_type; } // handle command packet. if(AV_RL32(mms->in_buffer + 4)==0xb00bface) { int length_remaining, hr; mms->incoming_flags= mms->in_buffer[3]; read_result= url_read_complete(mms->mms_hd, mms->in_buffer+8, 4); if(read_result != 4) { av_log(NULL, AV_LOG_ERROR, "Reading command packet length failed: %d (%s)/n", read_result, read_result < 0 ? strerror(read_result) : "The server closed the connection"); return read_result < 0 ? read_result : AVERROR_IO; } length_remaining= AV_RL32(mms->in_buffer+8) + 4; dprintf(NULL, "Length remaining is %d/n", length_remaining); // read the rest of the packet. if (length_remaining < 0 || length_remaining > sizeof(mms->in_buffer) - 12) { av_log(NULL, AV_LOG_ERROR, "Incoming packet length %d exceeds bufsize %zu/n", length_remaining, sizeof(mms->in_buffer) - 12); return AVERROR_INVALIDDATA; } read_result = url_read_complete(mms->mms_hd, mms->in_buffer + 12, length_remaining) ; if (read_result != length_remaining) { av_log(NULL, AV_LOG_ERROR, "Reading pkt data (length=%d) failed: %d (%s)/n", length_remaining, read_result, read_result < 0 ? strerror(read_result) : "The server closed the connection"); return read_result < 0 ? read_result : AVERROR_IO; } packet_type= AV_RL16(mms->in_buffer+36); hr = AV_RL32(mms->in_buffer + 40); if (hr) { av_log(NULL, AV_LOG_ERROR, "Server sent an error status code: 0x%08x/n", hr); return AVERROR_UNKNOWN; } } else { int length_remaining; int packet_id_type; int tmp; // note we cache the first 8 bytes, // then fill up the buffer with the others tmp = AV_RL16(mms->in_buffer + 6); length_remaining = (tmp - 8) & 0xffff; mms->incoming_packet_seq = AV_RL32(mms->in_buffer); packet_id_type = mms->in_buffer[4]; mms->incoming_flags = mms->in_buffer[5]; if (length_remaining < 0 || length_remaining > sizeof(mms->in_buffer) - 8) { av_log(NULL, AV_LOG_ERROR, "Data length %d is invalid or too large (max=%zu)/n", length_remaining, sizeof(mms->in_buffer)); return AVERROR_INVALIDDATA; } mms->remaining_in_len = length_remaining; mms->read_in_ptr = mms->in_buffer; read_result= url_read_complete(mms->mms_hd, mms->in_buffer, length_remaining); if(read_result != length_remaining) { av_log(NULL, AV_LOG_ERROR, "Failed to read packet data of size %d: %d (%s)/n", length_remaining, read_result, read_result < 0 ? strerror(read_result) : "The server closed the connection"); return read_result < 0 ? read_result : AVERROR_IO; } // if we successfully read everything. if(packet_id_type == mms->header_packet_id) { packet_type = SC_PKT_ASF_HEADER; // Store the asf header//.........这里部分代码省略.........
开发者ID:Akuaksh，项目名称:FFmpeg-alsenc，代码行数:101，
示例11: wv_read_block_headerstatic int wv_read_block_header(AVFormatContext *ctx, ByteIOContext *pb){ WVContext *wc = ctx->priv_data; uint32_t tag, ver; int size; int rate, bpp, chan; wc->pos = url_ftell(pb); tag = get_le32(pb); if (tag != MKTAG('w', 'v', 'p', 'k')) return -1; size = get_le32(pb); if(size < 24 || size > WV_BLOCK_LIMIT){ av_log(ctx, AV_LOG_ERROR, "Incorrect block size %i/n", size); return -1; } wc->blksize = size; ver = get_le16(pb); if(ver < 0x402 || ver > 0x410){ av_log(ctx, AV_LOG_ERROR, "Unsupported version %03X/n", ver); return -1; } get_byte(pb); // track no get_byte(pb); // track sub index wc->samples = get_le32(pb); // total samples in file wc->soff = get_le32(pb); // offset in samples of current block get_buffer(pb, wc->extra, WV_EXTRA_SIZE); wc->flags = AV_RL32(wc->extra + 4); //parse flags if(wc->flags & WV_FLOAT){ av_log(ctx, AV_LOG_ERROR, "Floating point data is not supported/n"); return -1; } if(wc->flags & WV_HYBRID){ av_log(ctx, AV_LOG_ERROR, "Hybrid coding mode is not supported/n"); return -1; } bpp = ((wc->flags & 3) + 1) << 3; chan = 1 + !(wc->flags & WV_MONO); rate = wv_rates[(wc->flags >> 23) & 0xF]; if(rate == -1){ av_log(ctx, AV_LOG_ERROR, "Unknown sampling rate/n"); return -1; } if(!wc->bpp) wc->bpp = bpp; if(!wc->chan) wc->chan = chan; if(!wc->rate) wc->rate = rate; if(wc->flags && bpp != wc->bpp){ av_log(ctx, AV_LOG_ERROR, "Bits per sample differ, this block: %i, header block: %i/n", bpp, wc->bpp); return -1; } if(wc->flags && chan != wc->chan){ av_log(ctx, AV_LOG_ERROR, "Channels differ, this block: %i, header block: %i/n", chan, wc->chan); return -1; } if(wc->flags && rate != wc->rate){ av_log(ctx, AV_LOG_ERROR, "Sampling rate differ, this block: %i, header block: %i/n", rate, wc->rate); return -1; } wc->blksize = size - 24; return 0;}
开发者ID:MichaelH13，项目名称:sdkpub，代码行数:64，
示例12: read_packetstatic int read_packet(AVFormatContext *s, AVPacket *pkt){ BinkDemuxContext *bink = s->priv_data; AVIOContext *pb = s->pb; int ret; if (bink->current_track < 0) { int index_entry; AVStream *st = s->streams[0]; // stream 0 is video stream with index if (bink->video_pts >= st->duration) return AVERROR(EIO); index_entry = av_index_search_timestamp(st, bink->video_pts, AVSEEK_FLAG_ANY); if (index_entry < 0) { av_log(s, AV_LOG_ERROR, "could not find index entry for frame %"PRId64"/n", bink->video_pts); return AVERROR(EIO); } bink->remain_packet_size = st->index_entries[index_entry].size; bink->current_track = 0; } while (bink->current_track < bink->num_audio_tracks) { uint32_t audio_size = avio_rl32(pb); if (audio_size > bink->remain_packet_size - 4) { av_log(s, AV_LOG_ERROR, "frame %"PRId64": audio size in header (%u) > size of packet left (%u)/n", bink->video_pts, audio_size, bink->remain_packet_size); return AVERROR(EIO); } bink->remain_packet_size -= 4 + audio_size; bink->current_track++; if (audio_size >= 4) { /* get one audio packet per track */ if ((ret = av_get_packet(pb, pkt, audio_size)) < 0) return ret; pkt->stream_index = bink->current_track; pkt->pts = bink->audio_pts[bink->current_track - 1]; /* Each audio packet reports the number of decompressed samples (in bytes). We use this value to calcuate the audio PTS */ if (pkt->size >= 4) bink->audio_pts[bink->current_track -1] += AV_RL32(pkt->data) / (2 * s->streams[bink->current_track]->codec->channels); return 0; } else { avio_skip(pb, audio_size); } } /* get video packet */ if ((ret = av_get_packet(pb, pkt, bink->remain_packet_size)) < 0) return ret; pkt->stream_index = 0; pkt->pts = bink->video_pts++; pkt->flags |= AV_PKT_FLAG_KEY; /* -1 instructs the next call to read_packet() to read the next frame */ bink->current_track = -1; return 0;}
开发者ID:ares89，项目名称:vlc，代码行数:66，
示例13: decode_framestatic int decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *avpkt){ JvContext *s = avctx->priv_data; int buf_size = avpkt->size; const uint8_t *buf = avpkt->data; const uint8_t *buf_end = buf + buf_size; int video_size, video_type, i, j, ret; video_size = AV_RL32(buf); video_type = buf[4]; buf += 5; if (video_size) { if ((ret = ff_reget_buffer(avctx, s->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed/n"); return ret; } if (video_type == 0 || video_type == 1) { GetBitContext gb; init_get_bits(&gb, buf, 8 * FFMIN(video_size, buf_end - buf)); for (j = 0; j < avctx->height; j += 8) for (i = 0; i < avctx->width; i += 8) decode8x8(&gb, s->frame->data[0] + j * s->frame->linesize[0] + i, s->frame->linesize[0], &s->bdsp); buf += video_size; } else if (video_type == 2) { if (buf + 1 <= buf_end) { int v = *buf++; for (j = 0; j < avctx->height; j++) memset(s->frame->data[0] + j * s->frame->linesize[0], v, avctx->width); } } else { av_log(avctx, AV_LOG_WARNING, "unsupported frame type %i/n", video_type); return AVERROR_INVALIDDATA; } } if (buf < buf_end) { for (i = 0; i < AVPALETTE_COUNT && buf + 3 <= buf_end; i++) { s->palette[i] = AV_RB24(buf) << 2; buf += 3; } s->palette_has_changed = 1; } if (video_size) { s->frame->key_frame = 1; s->frame->pict_type = AV_PICTURE_TYPE_I; s->frame->palette_has_changed = s->palette_has_changed; s->palette_has_changed = 0; memcpy(s->frame->data[1], s->palette, AVPALETTE_SIZE); if ((ret = av_frame_ref(data, s->frame)) < 0) return ret; *got_frame = 1; } return buf_size;}
开发者ID:OS2World，项目名称:LIB-libav，代码行数:66，
示例14: write_packetstatic int write_packet(AVFormatContext *s, AVPacket *pkt){ VideoData *img = s->priv_data; AVIOContext *pb[3]; char filename[1024]; AVCodecContext *codec = s->streams[ pkt->stream_index ]->codec; int i; if (!img->is_pipe) { if (av_get_frame_filename(filename, sizeof(filename), img->path, img->img_number) < 0 && img->img_number > 1) { av_log(s, AV_LOG_ERROR, "Could not get frame filename number %d from pattern '%s'/n", img->img_number, img->path); return AVERROR(EIO); } for(i = 0; i < 3; i++) { if (avio_open(&pb[i], filename, AVIO_WRONLY) < 0) { av_log(s, AV_LOG_ERROR, "Could not open file : %s/n", filename); return AVERROR(EIO); } if(!img->split_planes) break; filename[ strlen(filename) - 1 ] = 'U' + i; } } else { pb[0] = s->pb; } if(img->split_planes) { int ysize = codec->width * codec->height; avio_write(pb[0], pkt->data , ysize); avio_write(pb[1], pkt->data + ysize, (pkt->size - ysize) / 2); avio_write(pb[2], pkt->data + ysize + (pkt->size - ysize) / 2, (pkt->size - ysize) / 2); avio_flush(pb[1]); avio_flush(pb[2]); avio_close(pb[1]); avio_close(pb[2]); } else { if(av_str2id(img_tags, s->filename) == CODEC_ID_JPEG2000) { AVStream *st = s->streams[0]; if(st->codec->extradata_size > 8 && AV_RL32(st->codec->extradata + 4) == MKTAG('j', 'p', '2', 'h')) { if(pkt->size < 8 || AV_RL32(pkt->data + 4) != MKTAG('j', 'p', '2', 'c')) goto error; avio_wb32(pb[0], 12); ffio_wfourcc(pb[0], "jP "); avio_wb32(pb[0], 0x0D0A870A); // signature avio_wb32(pb[0], 20); ffio_wfourcc(pb[0], "ftyp"); ffio_wfourcc(pb[0], "jp2 "); avio_wb32(pb[0], 0); ffio_wfourcc(pb[0], "jp2 "); avio_write(pb[0], st->codec->extradata, st->codec->extradata_size); } else if(pkt->size < 8 || (!st->codec->extradata_size && AV_RL32(pkt->data + 4) != MKTAG('j', 'P', ' ', ' '))) // signature {error: av_log(s, AV_LOG_ERROR, "malformated jpeg2000 codestream/n"); return -1; } } avio_write(pb[0], pkt->data, pkt->size); } avio_flush(pb[0]); if (!img->is_pipe) { avio_close(pb[0]); } img->img_number++; return 0;}
开发者ID:248668342，项目名称:ffmpeg-windows，代码行数:87，
示例15: roq_read_packetstatic int roq_read_packet(AVFormatContext *s, AVPacket *pkt){ RoqDemuxContext *roq = s->priv_data; AVIOContext *pb = s->pb; int ret = 0; unsigned int chunk_size; unsigned int chunk_type; unsigned int codebook_size; unsigned char preamble[RoQ_CHUNK_PREAMBLE_SIZE]; int packet_read = 0; int64_t codebook_offset; while (!packet_read) { if (avio_feof(s->pb)) return AVERROR(EIO); /* get the next chunk preamble */ if ((ret = avio_read(pb, preamble, RoQ_CHUNK_PREAMBLE_SIZE)) != RoQ_CHUNK_PREAMBLE_SIZE) return AVERROR(EIO); chunk_type = AV_RL16(&preamble[0]); chunk_size = AV_RL32(&preamble[2]); if(chunk_size > INT_MAX) return AVERROR_INVALIDDATA; chunk_size = ffio_limit(pb, chunk_size); switch (chunk_type) { case RoQ_INFO: if (roq->video_stream_index == -1) { AVStream *st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); avpriv_set_pts_info(st, 63, 1, roq->frame_rate); roq->video_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->codec->codec_id = AV_CODEC_ID_ROQ; st->codec->codec_tag = 0; /* no fourcc */ if (avio_read(pb, preamble, RoQ_CHUNK_PREAMBLE_SIZE) != RoQ_CHUNK_PREAMBLE_SIZE) return AVERROR(EIO); st->codec->width = roq->width = AV_RL16(preamble); st->codec->height = roq->height = AV_RL16(preamble + 2); break; } /* don't care about this chunk anymore */ avio_skip(pb, RoQ_CHUNK_PREAMBLE_SIZE); break; case RoQ_QUAD_CODEBOOK: if (roq->video_stream_index < 0) return AVERROR_INVALIDDATA; /* packet needs to contain both this codebook and next VQ chunk */ codebook_offset = avio_tell(pb) - RoQ_CHUNK_PREAMBLE_SIZE; codebook_size = chunk_size; avio_skip(pb, codebook_size); if (avio_read(pb, preamble, RoQ_CHUNK_PREAMBLE_SIZE) != RoQ_CHUNK_PREAMBLE_SIZE) return AVERROR(EIO); chunk_size = AV_RL32(&preamble[2]) + RoQ_CHUNK_PREAMBLE_SIZE * 2 + codebook_size; /* rewind */ avio_seek(pb, codebook_offset, SEEK_SET); /* load up the packet */ ret= av_get_packet(pb, pkt, chunk_size); if (ret != chunk_size) return AVERROR(EIO); pkt->stream_index = roq->video_stream_index; pkt->pts = roq->video_pts++; packet_read = 1; break; case RoQ_SOUND_MONO: case RoQ_SOUND_STEREO: if (roq->audio_stream_index == -1) { AVStream *st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); avpriv_set_pts_info(st, 32, 1, RoQ_AUDIO_SAMPLE_RATE); roq->audio_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = AV_CODEC_ID_ROQ_DPCM; st->codec->codec_tag = 0; /* no tag */ if (chunk_type == RoQ_SOUND_STEREO) { st->codec->channels = 2; st->codec->channel_layout = AV_CH_LAYOUT_STEREO; } else { st->codec->channels = 1; st->codec->channel_layout = AV_CH_LAYOUT_MONO; } roq->audio_channels = st->codec->channels; st->codec->sample_rate = RoQ_AUDIO_SAMPLE_RATE; st->codec->bits_per_coded_sample = 16;//.........这里部分代码省略.........
开发者ID:r-type，项目名称:vice-libretro，代码行数:101，
示例16: ea_read_packetstatic int ea_read_packet(AVFormatContext *s, AVPacket *pkt){ EaDemuxContext *ea = s->priv_data; AVIOContext *pb = s->pb; int partial_packet = 0; unsigned int chunk_type, chunk_size; int ret = 0, packet_read = 0, key = 0; int av_uninit(num_samples); while (!packet_read || partial_packet) { chunk_type = avio_rl32(pb); chunk_size = ea->big_endian ? avio_rb32(pb) : avio_rl32(pb); if (chunk_size <= 8) return AVERROR_INVALIDDATA; chunk_size -= 8; switch (chunk_type) { /* audio data */ case ISNh_TAG: /* header chunk also contains data; skip over the header portion */ if (chunk_size < 32) return AVERROR_INVALIDDATA; avio_skip(pb, 32); chunk_size -= 32; case ISNd_TAG: case SCDl_TAG: case SNDC_TAG: case SDEN_TAG: if (!ea->audio_codec) { avio_skip(pb, chunk_size); break; } else if (ea->audio_codec == AV_CODEC_ID_PCM_S16LE_PLANAR || ea->audio_codec == AV_CODEC_ID_MP3) { num_samples = avio_rl32(pb); avio_skip(pb, 8); chunk_size -= 12; } if (partial_packet) { avpriv_request_sample(s, "video header followed by audio packet"); av_free_packet(pkt); partial_packet = 0; } ret = av_get_packet(pb, pkt, chunk_size); if (ret < 0) return ret; pkt->stream_index = ea->audio_stream_index; switch (ea->audio_codec) { case AV_CODEC_ID_ADPCM_EA: case AV_CODEC_ID_ADPCM_EA_R1: case AV_CODEC_ID_ADPCM_EA_R2: case AV_CODEC_ID_ADPCM_IMA_EA_EACS: case AV_CODEC_ID_ADPCM_EA_R3: if (pkt->size < 4) { av_log(s, AV_LOG_ERROR, "Packet is too short/n"); av_free_packet(pkt); return AVERROR_INVALIDDATA; } if (ea->audio_codec == AV_CODEC_ID_ADPCM_EA_R3) pkt->duration = AV_RB32(pkt->data); else pkt->duration = AV_RL32(pkt->data); break; case AV_CODEC_ID_ADPCM_IMA_EA_SEAD: pkt->duration = ret * 2 / ea->num_channels; break; case AV_CODEC_ID_PCM_S16LE_PLANAR: case AV_CODEC_ID_MP3: pkt->duration = num_samples; break; default: pkt->duration = chunk_size / (ea->bytes * ea->num_channels); } packet_read = 1; break; /* ending tag */ case 0: case ISNe_TAG: case SCEl_TAG: case SEND_TAG: case SEEN_TAG: ret = AVERROR(EIO); packet_read = 1; break; case MVIh_TAG: case kVGT_TAG: case pQGT_TAG: case TGQs_TAG: case MADk_TAG: key = AV_PKT_FLAG_KEY; case MVIf_TAG: case fVGT_TAG: case MADm_TAG: case MADe_TAG: avio_seek(pb, -8, SEEK_CUR); // include chunk preamble chunk_size += 8; goto get_video_packet;//.........这里部分代码省略.........
开发者ID:AsamQi，项目名称:FFmpeg，代码行数:101，
示例17: flic_decode_initstatic av_cold int flic_decode_init(AVCodecContext *avctx){ FlicDecodeContext *s = avctx->priv_data; unsigned char *fli_header = (unsigned char *)avctx->extradata; int depth; if (avctx->extradata_size != 0 && avctx->extradata_size != 12 && avctx->extradata_size != 128 && avctx->extradata_size != 256 && avctx->extradata_size != 904 && avctx->extradata_size != 1024) { av_log(avctx, AV_LOG_ERROR, "Unexpected extradata size %d/n", avctx->extradata_size); return AVERROR_INVALIDDATA; } s->avctx = avctx; if (s->avctx->extradata_size == 12) { /* special case for magic carpet FLIs */ s->fli_type = FLC_MAGIC_CARPET_SYNTHETIC_TYPE_CODE; depth = 8; } else if (avctx->extradata_size == 1024) { uint8_t *ptr = avctx->extradata; int i; for (i = 0; i < 256; i++) { s->palette[i] = AV_RL32(ptr); ptr += 4; } depth = 8; /* FLI in MOV, see e.g. FFmpeg trac issue #626 */ } else if (avctx->extradata_size == 0 || avctx->extradata_size == 256 || /* see FFmpeg ticket #1234 */ avctx->extradata_size == 904) { s->fli_type = FLI_TYPE_CODE; depth = 8; } else { s->fli_type = AV_RL16(&fli_header[4]); depth = AV_RL16(&fli_header[12]); } if (depth == 0) { depth = 8; /* Some FLC generators set depth to zero, when they mean 8Bpp. Fix up here */ } if ((s->fli_type == FLC_FLX_TYPE_CODE) && (depth == 16)) { depth = 15; /* Original Autodesk FLX's say the depth is 16Bpp when it is really 15Bpp */ } switch (depth) { case 8 : avctx->pix_fmt = AV_PIX_FMT_PAL8; break; case 15 : avctx->pix_fmt = AV_PIX_FMT_RGB555; break; case 16 : avctx->pix_fmt = AV_PIX_FMT_RGB565; break; case 24 : avctx->pix_fmt = AV_PIX_FMT_BGR24; /* Supposedly BGR, but havent any files to test with */ avpriv_request_sample(avctx, "24Bpp FLC/FLX"); return AVERROR_PATCHWELCOME; default : av_log(avctx, AV_LOG_ERROR, "Unknown FLC/FLX depth of %d Bpp is unsupported./n",depth); return AVERROR_INVALIDDATA; } s->frame = av_frame_alloc(); if (!s->frame) return AVERROR(ENOMEM); s->new_palette = 0; return 0;}
开发者ID:0Soul，项目名称:FFmpeg，代码行数:71，
示例18: rm_read_audio_stream_infostatic int rm_read_audio_stream_info(AVFormatContext *s, AVIOContext *pb, AVStream *st, RMStream *ast, int read_all){ char buf[256]; uint32_t version; int ret; /* ra type header */ version = avio_rb16(pb); /* version */ if (version == 3) { unsigned bytes_per_minute; int header_size = avio_rb16(pb); int64_t startpos = avio_tell(pb); avio_skip(pb, 8); bytes_per_minute = avio_rb16(pb); avio_skip(pb, 4); rm_read_metadata(s, pb, 0); if ((startpos + header_size) >= avio_tell(pb) + 2) { // fourcc (should always be "lpcJ") avio_r8(pb); get_str8(pb, buf, sizeof(buf)); } // Skip extra header crap (this should never happen) if ((startpos + header_size) > avio_tell(pb)) avio_skip(pb, header_size + startpos - avio_tell(pb)); if (bytes_per_minute) st->codec->bit_rate = 8LL * bytes_per_minute / 60; st->codec->sample_rate = 8000; st->codec->channels = 1; st->codec->channel_layout = AV_CH_LAYOUT_MONO; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = AV_CODEC_ID_RA_144; ast->deint_id = DEINT_ID_INT0; } else { int flavor, sub_packet_h, coded_framesize, sub_packet_size; int codecdata_length; unsigned bytes_per_minute; /* old version (4) */ avio_skip(pb, 2); /* unused */ avio_rb32(pb); /* .ra4 */ avio_rb32(pb); /* data size */ avio_rb16(pb); /* version2 */ avio_rb32(pb); /* header size */ flavor= avio_rb16(pb); /* add codec info / flavor */ ast->coded_framesize = coded_framesize = avio_rb32(pb); /* coded frame size */ avio_rb32(pb); /* ??? */ bytes_per_minute = avio_rb32(pb); if (version == 4) { if (bytes_per_minute) st->codec->bit_rate = 8LL * bytes_per_minute / 60; } avio_rb32(pb); /* ??? */ ast->sub_packet_h = sub_packet_h = avio_rb16(pb); /* 1 */ st->codec->block_align= avio_rb16(pb); /* frame size */ ast->sub_packet_size = sub_packet_size = avio_rb16(pb); /* sub packet size */ avio_rb16(pb); /* ??? */ if (version == 5) { avio_rb16(pb); avio_rb16(pb); avio_rb16(pb); } st->codec->sample_rate = avio_rb16(pb); avio_rb32(pb); st->codec->channels = avio_rb16(pb); if (version == 5) { ast->deint_id = avio_rl32(pb); avio_read(pb, buf, 4); buf[4] = 0; } else { AV_WL32(buf, 0); get_str8(pb, buf, sizeof(buf)); /* desc */ ast->deint_id = AV_RL32(buf); get_str8(pb, buf, sizeof(buf)); /* desc */ } st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_tag = AV_RL32(buf); st->codec->codec_id = ff_codec_get_id(ff_rm_codec_tags, st->codec->codec_tag); switch (st->codec->codec_id) { case AV_CODEC_ID_AC3: st->need_parsing = AVSTREAM_PARSE_FULL; break; case AV_CODEC_ID_RA_288: st->codec->extradata_size= 0; ast->audio_framesize = st->codec->block_align; st->codec->block_align = coded_framesize; break; case AV_CODEC_ID_COOK: st->need_parsing = AVSTREAM_PARSE_HEADERS; case AV_CODEC_ID_ATRAC3: case AV_CODEC_ID_SIPR: if (read_all) { codecdata_length = 0; } else { avio_rb16(pb); avio_r8(pb); if (version == 5) avio_r8(pb); codecdata_length = avio_rb32(pb); if(codecdata_length + FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)codecdata_length){ av_log(s, AV_LOG_ERROR, "codecdata_length too large/n"); return -1;//.........这里部分代码省略.........
开发者ID:Acidburn0zzz，项目名称:FFmpeg，代码行数:101，
示例19: fourxm_read_packetstatic int fourxm_read_packet(AVFormatContext *s, AVPacket *pkt){ FourxmDemuxContext *fourxm = s->priv_data; AVIOContext *pb = s->pb; unsigned int fourcc_tag; unsigned int size; int ret = 0; unsigned int track_number; int packet_read = 0; unsigned char header[8]; int audio_frame_count; while (!packet_read) { if ((ret = avio_read(s->pb, header, 8)) < 0) return ret; fourcc_tag = AV_RL32(&header[0]); size = AV_RL32(&header[4]); if (url_feof(pb)) return AVERROR(EIO); switch (fourcc_tag) { case LIST_TAG: /* this is a good time to bump the video pts */ fourxm->video_pts ++; /* skip the LIST-* tag and move on to the next fourcc */ avio_rl32(pb); break; case ifrm_TAG: case pfrm_TAG: case cfrm_TAG: case ifr2_TAG: case pfr2_TAG: case cfr2_TAG: /* allocate 8 more bytes than 'size' to account for fourcc * and size */ if (size + 8 < size || av_new_packet(pkt, size + 8)) return AVERROR(EIO); pkt->stream_index = fourxm->video_stream_index; pkt->pts = fourxm->video_pts; pkt->pos = avio_tell(s->pb); memcpy(pkt->data, header, 8); ret = avio_read(s->pb, &pkt->data[8], size); if (ret < 0){ av_free_packet(pkt); }else packet_read = 1; break; case snd__TAG: track_number = avio_rl32(pb); avio_skip(pb, 4); size-=8; if (track_number < fourxm->track_count && fourxm->tracks[track_number].channels>0) { ret= av_get_packet(s->pb, pkt, size); if(ret<0) return AVERROR(EIO); pkt->stream_index = fourxm->tracks[track_number].stream_index; pkt->pts = fourxm->tracks[track_number].audio_pts; packet_read = 1; /* pts accounting */ audio_frame_count = size; if (fourxm->tracks[track_number].adpcm) audio_frame_count -= 2 * (fourxm->tracks[track_number].channels); audio_frame_count /= fourxm->tracks[track_number].channels; if (fourxm->tracks[track_number].adpcm){ audio_frame_count *= 2; }else audio_frame_count /= (fourxm->tracks[track_number].bits / 8); fourxm->tracks[track_number].audio_pts += audio_frame_count; } else { avio_skip(pb, size); } break; default: avio_skip(pb, size); break; } } return ret;}
开发者ID:Brhett，项目名称:FFmpeg，代码行数:93，
示例20: ff_rm_read_mdpr_codecdataint ff_rm_read_mdpr_codecdata(AVFormatContext *s, AVIOContext *pb, AVStream *st, RMStream *rst, unsigned int codec_data_size, const uint8_t *mime){ unsigned int v; int size; int64_t codec_pos; int ret; if (codec_data_size > INT_MAX) return AVERROR_INVALIDDATA; avpriv_set_pts_info(st, 64, 1, 1000); codec_pos = avio_tell(pb); v = avio_rb32(pb); if (v == MKBETAG('M', 'L', 'T', 'I')) { int number_of_streams = avio_rb16(pb); int number_of_mdpr; int i; for (i = 0; i<number_of_streams; i++) avio_rb16(pb); number_of_mdpr = avio_rb16(pb); if (number_of_mdpr != 1) { avpriv_request_sample(s, "MLTI with multiple MDPR"); } avio_rb32(pb); v = avio_rb32(pb); } if (v == MKTAG(0xfd, 'a', 'r', '.')) { /* ra type header */ if (rm_read_audio_stream_info(s, pb, st, rst, 0)) return -1; } else if (v == MKBETAG('L', 'S', 'D', ':')) { avio_seek(pb, -4, SEEK_CUR); if ((ret = rm_read_extradata(pb, st->codec, codec_data_size)) < 0) return ret; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_tag = AV_RL32(st->codec->extradata); st->codec->codec_id = ff_codec_get_id(ff_rm_codec_tags, st->codec->codec_tag); } else if(mime && !strcmp(mime, "logical-fileinfo")){ int stream_count, rule_count, property_count, i; ff_free_stream(s, st); if (avio_rb16(pb) != 0) { av_log(s, AV_LOG_WARNING, "Unsupported version/n"); goto skip; } stream_count = avio_rb16(pb); avio_skip(pb, 6*stream_count); rule_count = avio_rb16(pb); avio_skip(pb, 2*rule_count); property_count = avio_rb16(pb); for(i=0; i<property_count; i++){ uint8_t name[128], val[128]; avio_rb32(pb); if (avio_rb16(pb) != 0) { av_log(s, AV_LOG_WARNING, "Unsupported Name value property version/n"); goto skip; //FIXME skip just this one } get_str8(pb, name, sizeof(name)); switch(avio_rb32(pb)) { case 2: get_strl(pb, val, sizeof(val), avio_rb16(pb)); av_dict_set(&s->metadata, name, val, 0); break; default: avio_skip(pb, avio_rb16(pb)); } } } else { int fps; if (avio_rl32(pb) != MKTAG('V', 'I', 'D', 'O')) { fail1: av_log(s, AV_LOG_WARNING, "Unsupported stream type %08x/n", v); goto skip; } st->codec->codec_tag = avio_rl32(pb); st->codec->codec_id = ff_codec_get_id(ff_rm_codec_tags, st->codec->codec_tag); av_dlog(s, "%X %X/n", st->codec->codec_tag, MKTAG('R', 'V', '2', '0')); if (st->codec->codec_id == AV_CODEC_ID_NONE) goto fail1; st->codec->width = avio_rb16(pb); st->codec->height = avio_rb16(pb); avio_skip(pb, 2); // looks like bits per sample avio_skip(pb, 4); // always zero? st->codec->codec_type = AVMEDIA_TYPE_VIDEO; st->need_parsing = AVSTREAM_PARSE_TIMESTAMPS; fps = avio_rb32(pb); if ((ret = rm_read_extradata(pb, st->codec, codec_data_size - (avio_tell(pb) - codec_pos))) < 0) return ret; if (fps > 0) { av_reduce(&st->avg_frame_rate.den, &st->avg_frame_rate.num, 0x10000, fps, (1 << 30) - 1);#if FF_API_R_FRAME_RATE st->r_frame_rate = st->avg_frame_rate;#endif//.........这里部分代码省略.........
开发者ID:Acidburn0zzz，项目名称:FFmpeg，代码行数:101，
示例21: flic_decode_frame_8BPPstatic int flic_decode_frame_8BPP(AVCodecContext *avctx, void *data, int *data_size, const uint8_t *buf, int buf_size){ FlicDecodeContext *s = avctx->priv_data; int stream_ptr = 0; int stream_ptr_after_color_chunk; int pixel_ptr; int palette_ptr; unsigned char palette_idx1; unsigned char palette_idx2; unsigned int frame_size; int num_chunks; unsigned int chunk_size; int chunk_type; int i, j; int color_packets; int color_changes; int color_shift; unsigned char r, g, b; int lines; int compressed_lines; int starting_line; signed short line_packets; int y_ptr; int byte_run; int pixel_skip; int pixel_countdown; unsigned char *pixels; unsigned int pixel_limit; s->frame.reference = 1; s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE; if (avctx->reget_buffer(avctx, &s->frame) < 0) { av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed/n"); return -1; } pixels = s->frame.data[0]; pixel_limit = s->avctx->height * s->frame.linesize[0]; frame_size = AV_RL32(&buf[stream_ptr]); stream_ptr += 6; /* skip the magic number */ num_chunks = AV_RL16(&buf[stream_ptr]); stream_ptr += 10; /* skip padding */ frame_size -= 16; /* iterate through the chunks */ while ((frame_size > 0) && (num_chunks > 0)) { chunk_size = AV_RL32(&buf[stream_ptr]); if (chunk_size > frame_size) { av_log(avctx, AV_LOG_WARNING, "Invalid chunk_size = %u > frame_size = %u/n", chunk_size, frame_size); chunk_size = frame_size; } stream_ptr += 4; chunk_type = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; switch (chunk_type) { case FLI_256_COLOR: case FLI_COLOR: stream_ptr_after_color_chunk = stream_ptr + chunk_size - 6; /* check special case: If this file is from the Magic Carpet * game and uses 6-bit colors even though it reports 256-color * chunks in a 0xAF12-type file (fli_type is set to 0xAF13 during * initialization) */ if ((chunk_type == FLI_256_COLOR) && (s->fli_type != FLC_MAGIC_CARPET_SYNTHETIC_TYPE_CODE)) color_shift = 0; else color_shift = 2; /* set up the palette */ color_packets = AV_RL16(&buf[stream_ptr]); stream_ptr += 2; palette_ptr = 0; for (i = 0; i < color_packets; i++) { /* first byte is how many colors to skip */ palette_ptr += buf[stream_ptr++]; /* next byte indicates how many entries to change */ color_changes = buf[stream_ptr++]; /* if there are 0 color changes, there are actually 256 */ if (color_changes == 0) color_changes = 256; for (j = 0; j < color_changes; j++) { unsigned int entry; /* wrap around, for good measure */ if ((unsigned)palette_ptr >= 256) palette_ptr = 0;//.........这里部分代码省略.........
开发者ID:Unhelpful，项目名称:ffmpeg，代码行数:101，
示例22: sox_probestatic int sox_probe(AVProbeData *p){ if (AV_RL32(p->buf) == SOX_TAG || AV_RB32(p->buf) == SOX_TAG) return AVPROBE_SCORE_MAX; return 0;}
开发者ID:eugenehp，项目名称:ffmbc，代码行数:6，
示例23: nuv_packetstatic int nuv_packet(AVFormatContext *s, AVPacket *pkt){ NUVContext *ctx = s->priv_data; AVIOContext *pb = s->pb; uint8_t hdr[HDRSIZE]; nuv_frametype frametype; int ret, size; while (!avio_feof(pb)) { int copyhdrsize = ctx->rtjpg_video ? HDRSIZE : 0; uint64_t pos = avio_tell(pb); ret = avio_read(pb, hdr, HDRSIZE); if (ret < HDRSIZE) return ret < 0 ? ret : AVERROR(EIO); frametype = hdr[0]; size = PKTSIZE(AV_RL32(&hdr[8])); switch (frametype) { case NUV_EXTRADATA: if (!ctx->rtjpg_video) { avio_skip(pb, size); break; } case NUV_VIDEO: if (ctx->v_id < 0) { av_log(s, AV_LOG_ERROR, "Video packet in file without video stream!/n"); avio_skip(pb, size); break; } ret = av_new_packet(pkt, copyhdrsize + size); if (ret < 0) return ret; pkt->pos = pos; pkt->flags |= hdr[2] == 0 ? AV_PKT_FLAG_KEY : 0; pkt->pts = AV_RL32(&hdr[4]); pkt->stream_index = ctx->v_id; memcpy(pkt->data, hdr, copyhdrsize); ret = avio_read(pb, pkt->data + copyhdrsize, size); if (ret < 0) { av_free_packet(pkt); return ret; } if (ret < size) av_shrink_packet(pkt, copyhdrsize + ret); return 0; case NUV_AUDIO: if (ctx->a_id < 0) { av_log(s, AV_LOG_ERROR, "Audio packet in file without audio stream!/n"); avio_skip(pb, size); break; } ret = av_get_packet(pb, pkt, size); pkt->flags |= AV_PKT_FLAG_KEY; pkt->pos = pos; pkt->pts = AV_RL32(&hdr[4]); pkt->stream_index = ctx->a_id; if (ret < 0) return ret; return 0; case NUV_SEEKP: // contains no data, size value is invalid break; default: avio_skip(pb, size); break; } } return AVERROR(EIO);}
开发者ID:markjreed，项目名称:vice-emu，代码行数:73，
示例24: aasc_decode_framestatic int aasc_decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *avpkt){ const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AascContext *s = avctx->priv_data; int compr, i, stride, psize, ret; if (buf_size < 4) { av_log(avctx, AV_LOG_ERROR, "frame too short/n"); return AVERROR_INVALIDDATA; } if ((ret = ff_reget_buffer(avctx, s->frame)) < 0) return ret; compr = AV_RL32(buf); buf += 4; buf_size -= 4; psize = avctx->bits_per_coded_sample / 8; switch (avctx->codec_tag) { case MKTAG('A', 'A', 'S', '4'): bytestream2_init(&s->gb, buf - 4, buf_size + 4); ff_msrle_decode(avctx, (AVPicture*)s->frame, 8, &s->gb); break; case MKTAG('A', 'A', 'S', 'C'): switch (compr) { case 0: stride = (avctx->width * psize + psize) & ~psize; if (buf_size < stride * avctx->height) return AVERROR_INVALIDDATA; for (i = avctx->height - 1; i >= 0; i--) { memcpy(s->frame->data[0] + i * s->frame->linesize[0], buf, avctx->width * psize); buf += stride; buf_size -= stride; } break; case 1: bytestream2_init(&s->gb, buf, buf_size); ff_msrle_decode(avctx, (AVPicture*)s->frame, 8, &s->gb); break; default: av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d/n", compr); return AVERROR_INVALIDDATA; } break; default: av_log(avctx, AV_LOG_ERROR, "Unknown FourCC: %X/n", avctx->codec_tag); return -1; } if (avctx->pix_fmt == AV_PIX_FMT_PAL8) memcpy(s->frame->data[1], s->palette, s->palette_size); *got_frame = 1; if ((ret = av_frame_ref(data, s->frame)) < 0) return ret; /* report that the buffer was completely consumed */ return buf_size;}
开发者ID:markjreed，项目名称:vice-emu，代码行数:62，
示例25: av_init_packetSTDMETHODIMP CDecAvcodec::Decode(const BYTE *buffer, int buflen, REFERENCE_TIME rtStartIn, REFERENCE_TIME rtStopIn, BOOL bSyncPoint, BOOL bDiscontinuity){ int got_picture = 0; int used_bytes = 0; BOOL bParserFrame = FALSE; 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_H264) { BOOL bRecovered = m_h264RandomAccess.searchRecoveryPoint(pDataBuffer, buflen); if (!bRecovered) { return S_OK; } } else 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; }//.........这里部分代码省略.........
开发者ID:cynics，项目名称:LAVFilters，代码行数:101，
示例26: ea_read_packetstatic int ea_read_packet(AVFormatContext *s, AVPacket *pkt){ EaDemuxContext *ea = s->priv_data; AVIOContext *pb = s->pb; int ret = 0; int packet_read = 0; unsigned int chunk_type, chunk_size; int key = 0; int num_samples = 0; while (!packet_read) { chunk_type = avio_rl32(pb); chunk_size = (ea->big_endian ? avio_rb32(pb) : avio_rl32(pb)) - 8; switch (chunk_type) { /* audio data */ case ISNh_TAG: /* header chunk also contains data; skip over the header portion*/ avio_skip(pb, 32); chunk_size -= 32; case ISNd_TAG: case SCDl_TAG: case SNDC_TAG: case SDEN_TAG: if (!ea->audio_codec) { avio_skip(pb, chunk_size); break; } else if (ea->audio_codec == CODEC_ID_PCM_S16LE_PLANAR || ea->audio_codec == CODEC_ID_MP3) { num_samples = avio_rl32(pb); avio_skip(pb, 8); chunk_size -= 12; } ret = av_get_packet(pb, pkt, chunk_size); if (ret < 0) return ret; if (ea->audio_codec == CODEC_ID_ADPCM_EA_R1 || ea->audio_codec == CODEC_ID_ADPCM_EA_R2 || ea->audio_codec == CODEC_ID_ADPCM_EA_R3) { num_samples = ea->audio_codec == CODEC_ID_ADPCM_EA_R3 ? AV_RB32(pkt->data) : AV_RL32(pkt->data); } pkt->stream_index = ea->audio_stream_index; pkt->pts = ea->audio_frame_counter; switch (ea->audio_codec) { case CODEC_ID_ADPCM_EA: /* 2 samples/byte, 1 or 2 samples per frame depending * on stereo; chunk also has 12-byte header */ ea->audio_frame_counter += ((chunk_size - 12) * 2) / ea->num_channels; break; default: if (num_samples > 0) ea->audio_frame_counter += num_samples; else ea->audio_frame_counter += chunk_size / (ea->bytes * ea->num_channels); } packet_read = 1; break; /* ending tag */ case 0: case ISNe_TAG: case SCEl_TAG: case SEND_TAG: case SEEN_TAG: ret = AVERROR(EIO); packet_read = 1; break; case MVIh_TAG: case kVGT_TAG: case pQGT_TAG: case TGQs_TAG: case MADk_TAG: key = AV_PKT_FLAG_KEY; case MVIf_TAG: case fVGT_TAG: case MADm_TAG: case MADe_TAG: avio_seek(pb, -8, SEEK_CUR); // include chunk preamble chunk_size += 8; goto get_video_packet; case mTCD_TAG: avio_skip(pb, 8); // skip ea dct header chunk_size -= 8; goto get_video_packet; case MV0K_TAG: case MPCh_TAG: case pIQT_TAG: key = AV_PKT_FLAG_KEY;//.........这里部分代码省略.........
开发者ID:eugenehp，项目名称:ffmbc，代码行数:101，
示例27: str_probestatic int str_probe(AVProbeData *p){ uint8_t *sector= p->buf; uint8_t *end= sector + p->buf_size; int aud=0, vid=0; if (p->buf_size < RAW_CD_SECTOR_SIZE) return 0; if ((AV_RL32(&p->buf[0]) == RIFF_TAG) && (AV_RL32(&p->buf[8]) == CDXA_TAG)) { /* RIFF header seen; skip 0x2C bytes */ sector += RIFF_HEADER_SIZE; } while (end - sector >= RAW_CD_SECTOR_SIZE) { /* look for CD sync header (00, 0xFF x 10, 00) */ if (memcmp(sector,sync_header,sizeof(sync_header))) return 0; if (sector[0x11] >= 32) return 0; switch (sector[0x12] & CDXA_TYPE_MASK) { case CDXA_TYPE_DATA: case CDXA_TYPE_VIDEO: { int current_sector = AV_RL16(§or[0x1C]); int sector_count = AV_RL16(§or[0x1E]); int frame_size = AV_RL32(§or[0x24]); if(!( frame_size>=0 && current_sector < sector_count && sector_count*VIDEO_DATA_CHUNK_SIZE >=frame_size)){ return 0; } /*st->codec->width = AV_RL16(§or[0x28]); st->codec->height = AV_RL16(§or[0x2A]);*/// if (current_sector == sector_count-1) { vid++;// } } break; case CDXA_TYPE_AUDIO: if(sector[0x13]&0x2A) return 0; aud++; break; default: if(sector[0x12] & CDXA_TYPE_MASK) return 0; } sector += RAW_CD_SECTOR_SIZE; } /* MPEG files (like those ripped from VCDs) can also look like this; * only return half certainty */ if(vid+aud > 3) return 50; else if(vid+aud) return 1; else return 0;}
开发者ID:KindDragon，项目名称:FFmpeg，代码行数:63，
示例28: tta_read_headerstatic int tta_read_header(AVFormatContext *s, AVFormatParameters *ap){ TTAContext *c = s->priv_data; AVStream *st; int i, channels, bps, samplerate, datalen, framelen; uint64_t framepos, start_offset; if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) ff_id3v1_read(s); start_offset = avio_tell(s->pb); if (avio_rl32(s->pb) != AV_RL32("TTA1")) return -1; // not tta file avio_skip(s->pb, 2); // FIXME: flags channels = avio_rl16(s->pb); bps = avio_rl16(s->pb); samplerate = avio_rl32(s->pb); if(samplerate <= 0 || samplerate > 1000000){ av_log(s, AV_LOG_ERROR, "nonsense samplerate/n"); return -1; } datalen = avio_rl32(s->pb); if(datalen < 0){ av_log(s, AV_LOG_ERROR, "nonsense datalen/n"); return -1; } avio_skip(s->pb, 4); // header crc framelen = samplerate*256/245; c->totalframes = datalen / framelen + ((datalen % framelen) ? 1 : 0); c->currentframe = 0; if(c->totalframes >= UINT_MAX/sizeof(uint32_t)){ av_log(s, AV_LOG_ERROR, "totalframes too large/n"); return -1; } st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); av_set_pts_info(st, 64, 1, samplerate); st->start_time = 0; st->duration = datalen; framepos = avio_tell(s->pb) + 4*c->totalframes + 4; for (i = 0; i < c->totalframes; i++) { uint32_t size = avio_rl32(s->pb); av_add_index_entry(st, framepos, i*framelen, size, 0, AVINDEX_KEYFRAME); framepos += size; } avio_skip(s->pb, 4); // seektable crc st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_TTA; st->codec->channels = channels; st->codec->sample_rate = samplerate; st->codec->bits_per_coded_sample = bps; st->codec->extradata_size = avio_tell(s->pb) - start_offset; if(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)st->codec->extradata_size){ //this check is redundant as avio_read should fail av_log(s, AV_LOG_ERROR, "extradata_size too large/n"); return -1; } st->codec->extradata = av_mallocz(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE); avio_seek(s->pb, start_offset, SEEK_SET); avio_read(s->pb, st->codec->extradata, st->codec->extradata_size); return 0;}
开发者ID:Tonyton，项目名称:gstreamer-ducati，代码行数:75，
示例29: process_ipmovie_chunk//.........这里部分代码省略......... chunk_size -= opcode_size; if (chunk_size < 0) { debug_ipmovie("chunk_size countdown just went negative/n"); chunk_type = CHUNK_BAD; break; } debug_ipmovie(" opcode type %02X, version %d, 0x%04X bytes: ", opcode_type, opcode_version, opcode_size); switch (opcode_type) { case OPCODE_END_OF_STREAM: debug_ipmovie("end of stream/n"); url_fseek(pb, opcode_size, SEEK_CUR); break; case OPCODE_END_OF_CHUNK: debug_ipmovie("end of chunk/n"); url_fseek(pb, opcode_size, SEEK_CUR); break; case OPCODE_CREATE_TIMER: debug_ipmovie("create timer/n"); if ((opcode_version > 0) || (opcode_size > 6)) { debug_ipmovie("bad create_timer opcode/n"); chunk_type = CHUNK_BAD; break; } if (get_buffer(pb, scratch, opcode_size) != opcode_size) { chunk_type = CHUNK_BAD; break; } s->fps = 1000000.0 / (AV_RL32(&scratch[0]) * AV_RL16(&scratch[4])); s->frame_pts_inc = 90000 / s->fps; debug_ipmovie(" %.2f frames/second (timer div = %d, subdiv = %d)/n", s->fps, AV_RL32(&scratch[0]), AV_RL16(&scratch[4])); break; case OPCODE_INIT_AUDIO_BUFFERS: debug_ipmovie("initialize audio buffers/n"); if ((opcode_version > 1) || (opcode_size > 10)) { debug_ipmovie("bad init_audio_buffers opcode/n"); chunk_type = CHUNK_BAD; break; } if (get_buffer(pb, scratch, opcode_size) != opcode_size) { chunk_type = CHUNK_BAD; break; } s->audio_sample_rate = AV_RL16(&scratch[4]); audio_flags = AV_RL16(&scratch[2]); /* bit 0 of the flags: 0 = mono, 1 = stereo */ s->audio_channels = (audio_flags & 1) + 1; /* bit 1 of the flags: 0 = 8 bit, 1 = 16 bit */ s->audio_bits = (((audio_flags >> 1) & 1) + 1) * 8; /* bit 2 indicates compressed audio in version 1 opcode */ if ((opcode_version == 1) && (audio_flags & 0x4)) s->audio_type = CODEC_ID_INTERPLAY_DPCM; else if (s->audio_bits == 16) s->audio_type = CODEC_ID_PCM_S16LE; else s->audio_type = CODEC_ID_PCM_U8; debug_ipmovie("audio: %d bits, %d Hz, %s, %s format/n", s->audio_bits,
开发者ID:apakian，项目名称:rtmp-cpp，代码行数:67，
示例30: wv_read_block_headerstatic int wv_read_block_header(AVFormatContext *ctx, AVIOContext *pb, int append){ WVContext *wc = ctx->priv_data; uint32_t tag, ver; int size; int rate, bpp, chan; uint32_t chmask; wc->pos = avio_tell(pb); if(!append){ tag = avio_rl32(pb); if (tag != MKTAG('w', 'v', 'p', 'k')) return -1; size = avio_rl32(pb); if(size < 24 || size > WV_BLOCK_LIMIT){ av_log(ctx, AV_LOG_ERROR, "Incorrect block size %i/n", size); return -1; } wc->blksize = size; ver = avio_rl16(pb); if(ver < 0x402 || ver > 0x410){ av_log(ctx, AV_LOG_ERROR, "Unsupported version %03X/n", ver); return -1; } avio_r8(pb); // track no avio_r8(pb); // track sub index wc->samples = avio_rl32(pb); // total samples in file wc->soff = avio_rl32(pb); // offset in samples of current block avio_read(pb, wc->extra, WV_EXTRA_SIZE); }else{ size = wc->blksize; } wc->flags = AV_RL32(wc->extra + 4); //parse flags bpp = ((wc->flags & 3) + 1) << 3; chan = 1 + !(wc->flags & WV_MONO); chmask = wc->flags & WV_MONO ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; rate = wv_rates[(wc->flags >> 23) & 0xF]; wc->multichannel = !!((wc->flags & WV_SINGLE_BLOCK) != WV_SINGLE_BLOCK); if(wc->multichannel){ chan = wc->chan; chmask = wc->chmask; } if((rate == -1 || !chan) && !wc->block_parsed){ int64_t block_end = avio_tell(pb) + wc->blksize - 24; if(!pb->seekable){ av_log(ctx, AV_LOG_ERROR, "Cannot determine additional parameters/n"); return -1; } while(avio_tell(pb) < block_end){ int id, size; id = avio_r8(pb); size = (id & 0x80) ? avio_rl24(pb) : avio_r8(pb); size <<= 1; if(id&0x40) size--; switch(id&0x3F){ case 0xD: if(size <= 1){ av_log(ctx, AV_LOG_ERROR, "Insufficient channel information/n"); return -1; } chan = avio_r8(pb); switch(size - 2){ case 0: chmask = avio_r8(pb); break; case 1: chmask = avio_rl16(pb); break; case 2: chmask = avio_rl24(pb); break; case 3: chmask = avio_rl32(pb); break; case 5: avio_skip(pb, 1); chan |= (avio_r8(pb) & 0xF) << 8; chmask = avio_rl24(pb); break; default: av_log(ctx, AV_LOG_ERROR, "Invalid channel info size %d/n", size); return -1; } break; case 0x27: rate = avio_rl24(pb); break; default: avio_skip(pb, size); } if(id&0x40) avio_skip(pb, 1); } if(rate == -1){ av_log(ctx, AV_LOG_ERROR, "Cannot determine custom sampling rate/n"); return -1; } avio_seek(pb, block_end - wc->blksize + 24, SEEK_SET);//.........这里部分代码省略.........
开发者ID:Amalerd，项目名称:SoxPlayer，代码行数:101，
注：本文中的AV_RL32函数示例整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。
C++ AV_WB16函数代码示例
C++ AV_RL16函数代码示例 |
