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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Encoding and processing into web-friendly formats
13 avril 2011, parMediaSPIP automatically converts uploaded files to internet-compatible formats.
Video files are encoded in MP4, Ogv and WebM (supported by HTML5) and MP4 (supported by Flash).
Audio files are encoded in MP3 and Ogg (supported by HTML5) and MP3 (supported by Flash).
Where possible, text is analyzed in order to retrieve the data needed for search engine detection, and then exported as a series of image files.
All uploaded files are stored online in their original format, so you can (...) -
Création définitive du canal
12 mars 2010, parLorsque votre demande est validée, vous pouvez alors procéder à la création proprement dite du canal. Chaque canal est un site à part entière placé sous votre responsabilité. Les administrateurs de la plateforme n’y ont aucun accès.
A la validation, vous recevez un email vous invitant donc à créer votre canal.
Pour ce faire il vous suffit de vous rendre à son adresse, dans notre exemple "http://votre_sous_domaine.mediaspip.net".
A ce moment là un mot de passe vous est demandé, il vous suffit d’y (...)
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How to fix a segmentaion fault in a C program ? [closed]
13 janvier 2012, par ipegasusPossible Duplicate :
Segmentation faultCurrently I am upgrading an open source program used for HTTP streaming. It needs to support the latest FFMPEG.
The code compiles fine with no warnings although I am getting a segmentation fault error.
I would like to know how to fix the issue ? and / or the best way to debug ? Please find attached a portion of the code due to size. I will try to add the project to github :) Thanks in advance !Sample Usage
# segmenter --i out.ts --l 10 --o stream.m3u8 --d segments --f streamMakefile
FFLIBS=`pkg-config --libs libavformat libavcodec libavutil`
FFFLAGS=`pkg-config --cflags libavformat libavcodec libavutil`
all:
gcc -Wall -g segmenter.c -o segmenter ${FFFLAGS} ${FFLIBS}segmenter.c
/*
* Copyright (c) 2009 Chase Douglas
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License version 2
* as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include
#include
#include
#include
#include
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include <sys></sys>stat.h>
#include "segmenter.h"
#include "libavformat/avformat.h"
#define IMAGE_ID3_SIZE 9171
void printUsage() {
fprintf(stderr, "\nExample: segmenter --i infile --d baseDir --f baseFileName --o playListFile.m3u8 --l 10 \n");
fprintf(stderr, "\nOptions: \n");
fprintf(stderr, "--i <infile>.\n");
fprintf(stderr, "--o <outfile>.\n");
fprintf(stderr, "--d basedir, the base directory for files.\n");
fprintf(stderr, "--f baseFileName, output files will be baseFileName-#.\n");
fprintf(stderr, "--l segment length, the length of each segment.\n");
fprintf(stderr, "--a, audio only decode for < 64k streams.\n");
fprintf(stderr, "--v, video only decode for < 64k streams.\n");
fprintf(stderr, "--version, print version details and exit.\n");
fprintf(stderr, "\n\n");
}
void ffmpeg_version() {
// output build and version numbers
fprintf(stderr, " libavutil version: %s\n", AV_STRINGIFY(LIBAVUTIL_VERSION));
fprintf(stderr, " libavutil build: %d\n", LIBAVUTIL_BUILD);
fprintf(stderr, " libavcodec version: %s\n", AV_STRINGIFY(LIBAVCODEC_VERSION));
fprintf(stdout, " libavcodec build: %d\n", LIBAVCODEC_BUILD);
fprintf(stderr, " libavformat version: %s\n", AV_STRINGIFY(LIBAVFORMAT_VERSION));
fprintf(stderr, " libavformat build: %d\n", LIBAVFORMAT_BUILD);
fprintf(stderr, " built on " __DATE__ " " __TIME__);
#ifdef __GNUC__
fprintf(stderr, ", gcc: " __VERSION__ "\n");
#else
fprintf(stderr, ", using a non-gcc compiler\n");
#endif
}
static AVStream *add_output_stream(AVFormatContext *output_format_context, AVStream *input_stream) {
AVCodecContext *input_codec_context;
AVCodecContext *output_codec_context;
AVStream *output_stream;
output_stream = avformat_new_stream(output_format_context, 0);
if (!output_stream) {
fprintf(stderr, "Segmenter error: Could not allocate stream\n");
exit(1);
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000) {
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
} else {
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type) {
#ifdef USE_OLD_FFMPEG
case CODEC_TYPE_AUDIO:
#else
case AVMEDIA_TYPE_AUDIO:
#endif
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3) {
output_codec_context->block_align = 0;
} else {
output_codec_context->block_align = input_codec_context->block_align;
}
break;
#ifdef USE_OLD_FFMPEG
case CODEC_TYPE_VIDEO:
#else
case AVMEDIA_TYPE_VIDEO:
#endif
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER) {
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
break;
default:
break;
}
return output_stream;
}
int write_index_file(const char index[], const char tmp_index[], const unsigned int planned_segment_duration, const unsigned int actual_segment_duration[],
const char output_directory[], const char output_prefix[], const char output_file_extension[],
const unsigned int first_segment, const unsigned int last_segment) {
FILE *index_fp;
char *write_buf;
unsigned int i;
index_fp = fopen(tmp_index, "w");
if (!index_fp) {
fprintf(stderr, "Could not open temporary m3u8 index file (%s), no index file will be created\n", tmp_index);
return -1;
}
write_buf = malloc(sizeof (char) * 1024);
if (!write_buf) {
fprintf(stderr, "Could not allocate write buffer for index file, index file will be invalid\n");
fclose(index_fp);
return -1;
}
unsigned int maxDuration = planned_segment_duration;
for (i = first_segment; i <= last_segment; i++)
if (actual_segment_duration[i] > maxDuration)
maxDuration = actual_segment_duration[i];
snprintf(write_buf, 1024, "#EXTM3U\n#EXT-X-TARGETDURATION:%u\n", maxDuration);
if (fwrite(write_buf, strlen(write_buf), 1, index_fp) != 1) {
fprintf(stderr, "Could not write to m3u8 index file, will not continue writing to index file\n");
free(write_buf);
fclose(index_fp);
return -1;
}
for (i = first_segment; i <= last_segment; i++) {
snprintf(write_buf, 1024, "#EXTINF:%u,\n%s-%u%s\n", actual_segment_duration[i], output_prefix, i, output_file_extension);
if (fwrite(write_buf, strlen(write_buf), 1, index_fp) != 1) {
fprintf(stderr, "Could not write to m3u8 index file, will not continue writing to index file\n");
free(write_buf);
fclose(index_fp);
return -1;
}
}
snprintf(write_buf, 1024, "#EXT-X-ENDLIST\n");
if (fwrite(write_buf, strlen(write_buf), 1, index_fp) != 1) {
fprintf(stderr, "Could not write last file and endlist tag to m3u8 index file\n");
free(write_buf);
fclose(index_fp);
return -1;
}
free(write_buf);
fclose(index_fp);
return rename(tmp_index, index);
}
int main(int argc, const char *argv[]) {
//input parameters
char inputFilename[MAX_FILENAME_LENGTH], playlistFilename[MAX_FILENAME_LENGTH], baseDirName[MAX_FILENAME_LENGTH], baseFileName[MAX_FILENAME_LENGTH];
char baseFileExtension[5]; //either "ts", "aac" or "mp3"
int segmentLength, outputStreams, verbosity, version;
char currentOutputFileName[MAX_FILENAME_LENGTH];
char tempPlaylistName[MAX_FILENAME_LENGTH];
//these are used to determine the exact length of the current segment
double prev_segment_time = 0;
double segment_time;
unsigned int actual_segment_durations[2048];
double packet_time = 0;
//new variables to keep track of output size
double output_bytes = 0;
unsigned int output_index = 1;
AVOutputFormat *ofmt;
AVFormatContext *ic = NULL;
AVFormatContext *oc;
AVStream *video_st = NULL;
AVStream *audio_st = NULL;
AVCodec *codec;
int video_index;
int audio_index;
unsigned int first_segment = 1;
unsigned int last_segment = 0;
int write_index = 1;
int decode_done;
int ret;
int i;
unsigned char id3_tag[128];
unsigned char * image_id3_tag;
size_t id3_tag_size = 73;
int newFile = 1; //a boolean value to flag when a new file needs id3 tag info in it
if (parseCommandLine(inputFilename, playlistFilename, baseDirName, baseFileName, baseFileExtension, &outputStreams, &segmentLength, &verbosity, &version, argc, argv) != 0)
return 0;
if (version) {
ffmpeg_version();
return 0;
}
fprintf(stderr, "%s %s\n", playlistFilename, tempPlaylistName);
image_id3_tag = malloc(IMAGE_ID3_SIZE);
if (outputStreams == OUTPUT_STREAM_AUDIO)
build_image_id3_tag(image_id3_tag);
build_id3_tag((char *) id3_tag, id3_tag_size);
snprintf(tempPlaylistName, strlen(playlistFilename) + strlen(baseDirName) + 1, "%s%s", baseDirName, playlistFilename);
strncpy(playlistFilename, tempPlaylistName, strlen(tempPlaylistName));
strncpy(tempPlaylistName, playlistFilename, MAX_FILENAME_LENGTH);
strncat(tempPlaylistName, ".", 1);
//decide if this is an aac file or a mpegts file.
//postpone deciding format until later
/* ifmt = av_find_input_format("mpegts");
if (!ifmt)
{
fprintf(stderr, "Could not find MPEG-TS demuxer.\n");
exit(1);
} */
av_log_set_level(AV_LOG_DEBUG);
av_register_all();
ret = avformat_open_input(&ic, inputFilename, NULL, NULL);
if (ret != 0) {
fprintf(stderr, "Could not open input file %s. Error %d.\n", inputFilename, ret);
exit(1);
}
if (avformat_find_stream_info(ic, NULL) < 0) {
fprintf(stderr, "Could not read stream information.\n");
exit(1);
}
oc = avformat_alloc_context();
if (!oc) {
fprintf(stderr, "Could not allocate output context.");
exit(1);
}
video_index = -1;
audio_index = -1;
for (i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++) {
switch (ic->streams[i]->codec->codec_type) {
#ifdef USE_OLD_FFMPEG
case CODEC_TYPE_VIDEO:
#else
case AVMEDIA_TYPE_VIDEO:
#endif
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
if (outputStreams & OUTPUT_STREAM_VIDEO)
video_st = add_output_stream(oc, ic->streams[i]);
break;
#ifdef USE_OLD_FFMPEG
case CODEC_TYPE_AUDIO:
#else
case AVMEDIA_TYPE_AUDIO:
#endif
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
if (outputStreams & OUTPUT_STREAM_AUDIO)
audio_st = add_output_stream(oc, ic->streams[i]);
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
if (video_index == -1) {
fprintf(stderr, "Stream must have video component.\n");
exit(1);
}
//now that we know the audio and video output streams
//we can decide on an output format.
if (outputStreams == OUTPUT_STREAM_AUDIO) {
//the audio output format should be the same as the audio input format
switch (ic->streams[audio_index]->codec->codec_id) {
case CODEC_ID_MP3:
fprintf(stderr, "Setting output audio to mp3.");
strncpy(baseFileExtension, ".mp3", strlen(".mp3"));
ofmt = av_guess_format("mp3", NULL, NULL);
break;
case CODEC_ID_AAC:
fprintf(stderr, "Setting output audio to aac.");
ofmt = av_guess_format("adts", NULL, NULL);
break;
default:
fprintf(stderr, "Codec id %d not supported.\n", ic->streams[audio_index]->id);
}
if (!ofmt) {
fprintf(stderr, "Could not find audio muxer.\n");
exit(1);
}
} else {
ofmt = av_guess_format("mpegts", NULL, NULL);
if (!ofmt) {
fprintf(stderr, "Could not find MPEG-TS muxer.\n");
exit(1);
}
}
oc->oformat = ofmt;
if (outputStreams & OUTPUT_STREAM_VIDEO && oc->oformat->flags & AVFMT_GLOBALHEADER) {
oc->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
/* Deprecated: pass the options to avformat_write_header directly.
if (av_set_parameters(oc, NULL) < 0) {
fprintf(stderr, "Invalid output format parameters.\n");
exit(1);
}
*/
av_dump_format(oc, 0, baseFileName, 1);
//open the video codec only if there is video data
if (video_index != -1) {
if (outputStreams & OUTPUT_STREAM_VIDEO)
codec = avcodec_find_decoder(video_st->codec->codec_id);
else
codec = avcodec_find_decoder(ic->streams[video_index]->codec->codec_id);
if (!codec) {
fprintf(stderr, "Could not find video decoder, key frames will not be honored.\n");
}
if (outputStreams & OUTPUT_STREAM_VIDEO)
ret = avcodec_open2(video_st->codec, codec, NULL);
else
avcodec_open2(ic->streams[video_index]->codec, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video decoder, key frames will not be honored.\n");
}
}
snprintf(currentOutputFileName, strlen(baseDirName) + strlen(baseFileName) + strlen(baseFileExtension) + 10, "%s%s-%u%s", baseDirName, baseFileName, output_index++, baseFileExtension);
if (avio_open(&oc->pb, currentOutputFileName, URL_WRONLY) < 0) {
fprintf(stderr, "Could not open '%s'.\n", currentOutputFileName);
exit(1);
}
newFile = 1;
int r = avformat_write_header(oc,NULL);
if (r) {
fprintf(stderr, "Could not write mpegts header to first output file.\n");
debugReturnCode(r);
exit(1);
}
//no segment info is written here. This just creates the shell of the playlist file
write_index = !write_index_file(playlistFilename, tempPlaylistName, segmentLength, actual_segment_durations, baseDirName, baseFileName, baseFileExtension, first_segment, last_segment);
do {
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0) {
break;
}
if (av_dup_packet(&packet) < 0) {
fprintf(stderr, "Could not duplicate packet.");
av_free_packet(&packet);
break;
}
//this time is used to check for a break in the segments
// if (packet.stream_index == video_index && (packet.flags & PKT_FLAG_KEY))
// {
// segment_time = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
// }
#if USE_OLD_FFMPEG
if (packet.stream_index == video_index && (packet.flags & PKT_FLAG_KEY))
#else
if (packet.stream_index == video_index && (packet.flags & AV_PKT_FLAG_KEY))
#endif
{
segment_time = (double) packet.pts * ic->streams[video_index]->time_base.num / ic->streams[video_index]->time_base.den;
}
// else if (video_index < 0)
// {
// segment_time = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
// }
//get the most recent packet time
//this time is used when the time for the final segment is printed. It may not be on the edge of
//of a keyframe!
if (packet.stream_index == video_index)
packet_time = (double) packet.pts * ic->streams[video_index]->time_base.num / ic->streams[video_index]->time_base.den; //(double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
else if (outputStreams & OUTPUT_STREAM_AUDIO)
packet_time = (double) audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
else
continue;
//start looking for segment splits for videos one half second before segment duration expires. This is because the
//segments are split on key frames so we cannot expect all segments to be split exactly equally.
if (segment_time - prev_segment_time >= segmentLength - 0.5) {
fprintf(stderr, "looking to print index file at time %lf\n", segment_time);
avio_flush(oc->pb);
avio_close(oc->pb);
if (write_index) {
actual_segment_durations[++last_segment] = (unsigned int) rint(segment_time - prev_segment_time);
write_index = !write_index_file(playlistFilename, tempPlaylistName, segmentLength, actual_segment_durations, baseDirName, baseFileName, baseFileExtension, first_segment, last_segment);
fprintf(stderr, "Writing index file at time %lf\n", packet_time);
}
struct stat st;
stat(currentOutputFileName, &st);
output_bytes += st.st_size;
snprintf(currentOutputFileName, strlen(baseDirName) + strlen(baseFileName) + strlen(baseFileExtension) + 10, "%s%s-%u%s", baseDirName, baseFileName, output_index++, baseFileExtension);
if (avio_open(&oc->pb, currentOutputFileName, URL_WRONLY) < 0) {
fprintf(stderr, "Could not open '%s'\n", currentOutputFileName);
break;
}
newFile = 1;
prev_segment_time = segment_time;
}
if (outputStreams == OUTPUT_STREAM_AUDIO && packet.stream_index == audio_index) {
if (newFile && outputStreams == OUTPUT_STREAM_AUDIO) {
//add id3 tag info
//fprintf(stderr, "adding id3tag to file %s\n", currentOutputFileName);
//printf("%lf %lld %lld %lld %lld %lld %lf\n", segment_time, audio_st->pts.val, audio_st->cur_dts, audio_st->cur_pkt.pts, packet.pts, packet.dts, packet.dts * av_q2d(ic->streams[audio_index]->time_base) );
fill_id3_tag((char*) id3_tag, id3_tag_size, packet.dts);
avio_write(oc->pb, id3_tag, id3_tag_size);
avio_write(oc->pb, image_id3_tag, IMAGE_ID3_SIZE);
avio_flush(oc->pb);
newFile = 0;
}
packet.stream_index = 0; //only one stream in audio only segments
ret = av_interleaved_write_frame(oc, &packet);
} else if (outputStreams & OUTPUT_STREAM_VIDEO) {
if (newFile) {
//fprintf(stderr, "New File: %lld %lld %lld\n", packet.pts, video_st->pts.val, audio_st->pts.val);
//printf("%lf %lld %lld %lld %lld %lld %lf\n", segment_time, audio_st->pts.val, audio_st->cur_dts, audio_st->cur_pkt.pts, packet.pts, packet.dts, packet.dts * av_q2d(ic->streams[audio_index]->time_base) );
newFile = 0;
}
if (outputStreams == OUTPUT_STREAM_VIDEO)
ret = av_write_frame(oc, &packet);
else
ret = av_interleaved_write_frame(oc, &packet);
}
if (ret < 0) {
fprintf(stderr, "Warning: Could not write frame of stream.\n");
} else if (ret > 0) {
fprintf(stderr, "End of stream requested.\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
} while (!decode_done);
//make sure all packets are written and then close the last file.
avio_flush(oc->pb);
av_write_trailer(oc);
if (video_st && video_st->codec)
avcodec_close(video_st->codec);
if (audio_st && audio_st->codec)
avcodec_close(audio_st->codec);
for (i = 0; i < oc->nb_streams; i++) {
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
avio_close(oc->pb);
av_free(oc);
struct stat st;
stat(currentOutputFileName, &st);
output_bytes += st.st_size;
if (write_index) {
actual_segment_durations[++last_segment] = (unsigned int) rint(packet_time - prev_segment_time);
//make sure that the last segment length is not zero
if (actual_segment_durations[last_segment] == 0)
actual_segment_durations[last_segment] = 1;
write_index_file(playlistFilename, tempPlaylistName, segmentLength, actual_segment_durations, baseDirName, baseFileName, baseFileExtension, first_segment, last_segment);
}
write_stream_size_file(baseDirName, baseFileName, output_bytes * 8 / segment_time);
return 0;
}
</outfile></infile> -
VP8 Codec Optimization Update
15 juin 2010, par noreply@blogger.com (John Luther) — inside webmSince WebM launched in May, the team has been working hard to make the VP8 video codec faster. Our community members have contributed improvements, but there’s more work to be done in some interesting areas related to performance (more on those below).
Encoder
The VP8 encoder is ripe for speed optimizations. Scott LaVarnway’s efforts in writing an x86 assembly version of the quantizer will help in this goal significantly as the quantizer is called many times while the encoder makes decisions about how much detail from the image will be transmitted.For those of you eager to get involved, one piece of low-hanging fruit is writing a SIMD version of the ARNR temporal filtering code. Also, much of the assembly code only makes use of the SSE2 instruction set, and there surely are newer extensions that could be made use of. There are also redundant code removal and other general cleanup to be done ; (Yaowu Xu has submitted some changes for these).
At a higher level, someone can explore some alternative motion search strategies in the encoder. Eventually the motion search can be decoupled entirely to allow motion fields to be calculated elsewhere (for example, on a graphics processor).Decoder
Decoder optimizations can bring higher resolutions and smoother playback to less powerful hardware.Jeff Muizelaar has submitted some changes which combine the IDCT and summation with the predicted block into a single function, helping us avoid storing the intermediate result, thus reducing memory transfers and avoiding cache pollution. This changes the assembly code in a fundamental way, so we will need to sync the other platforms up or switch them to a generic C implementation and accept the performance regression. Johann Koenig is working on implementing this change for ARM processors, and we’ll merge these changes into the mainline soon.
In addition, Tim Terriberry is attacking a different method of bounds checking on the "bool decoder." The bool decoder is performance-critical, as it is called several times for each bit in the input stream. The current code handles this check with a simple clamp in the innermost loops and a less-frequent copy into a circular buffer. This can be expensive at higher data rates. Tim’s patch removes the circular buffer, but uses a more complex clamp in the innermost loops. These inner loops have historically been troublesome on embedded platforms.
To contribute in these efforts, I’ve started working on rewriting higher-level parts of the decoder. I believe there is an opportunity to improve performance by paying better attention to data locality and cache layout, and reducing memory bus traffic in general. Another area I plan to explore is improving utilization in the multi-threaded decoder by separating the bitstream decoding from the rest of the image reconstruction, using work units larger than a single macroblock, and not tying functionality to a specific thread. To get involved in these areas, subscribe to the codec-devel mailing list and provide feedback on the code as it’s written.Embedded Processors
We want to optimize multiple platforms, not just desktops. Fritz Koenig has already started looking at the performance of VP8 on the Intel Atom platform. This platform need some attention as we wrote our current x86 assembly code with an out-of-order processor in mind. Since Atom is an in-order processor (much like the original Pentium), the instruction scheduling of all of the x86 assembly code needs to be reexamined. One option we’re looking at is scheduling the code for the Atom processor and seeing if that impacts the performance on other x86 platforms such as the Via C3 and AMD Geode. This is shaping up to be a lot of work, but doing it would provide us with an opportunity to tighten up our assembly code.
These issues, along with wanting to make better use of the larger register file on x86_64, may reignite every assembly programmer’s (least ?) favorite debate : whether or not to use intrinsics. Yunqing Wang has been experimenting with this a bit, but initial results aren’t promising. If you have experience in dealing with a lot of assembly code across several similar-but-kinda-different platforms, these maintainability issues might be familiar to you. I hope you’ll share your thoughts and experiences on the codec-devel mailing list.
Optimizing codecs is an iterative (some would say never-ending) process, so stay tuned for more posts on the progress we’re making, and by all means, start hacking yourself.
It’s exciting to see that we’re starting to get substantial code contributions from developers outside of Google, and I look forward to more as WebM grows into a strong community effort.
John Koleszar is a software engineer at Google. -
Legal Issues with FFMPEG ? [closed]
21 décembre 2011, par JimmeryHas anybody had any legal issues using FFMPEG in a commerical product, particularly from MPEG LA ?
I only ask from reading the bottom of this page : http://ffmpeg.org/legal.html
We are hoping to produce a website that uses FFMPEG (or similar program), currently the website will be free to use. However we do hope of monetizing aspects of the website later down the line, and wouldnt want to hit any legal issues at that point.
If this is an issue, does anybody know of a good alternative to FFMPEG which does not have these legal issues ?
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