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Autres articles (92)
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...) -
Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
While using MediaSPIP, you are invited to avoid using words like "Brand", "Cloud" and "Market".
MediaSPIP is designed to facilitate the sharing of creative media online, while allowing authors to retain complete control of their work.
MediaSPIP aims to be accessible to as many people as possible and development is based on expanding the (...) -
D’autres logiciels intéressants
12 avril 2011, parOn ne revendique pas d’être les seuls à faire ce que l’on fait ... et on ne revendique surtout pas d’être les meilleurs non plus ... Ce que l’on fait, on essaie juste de le faire bien, et de mieux en mieux...
La liste suivante correspond à des logiciels qui tendent peu ou prou à faire comme MediaSPIP ou que MediaSPIP tente peu ou prou à faire pareil, peu importe ...
On ne les connais pas, on ne les a pas essayé, mais vous pouvez peut être y jeter un coup d’oeil.
Videopress
Site Internet : (...)
Sur d’autres sites (6397)
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Write audio packet to file using ffmpeg
27 février 2017, par iamyzI am trying to write audio packet to file using ffmpeg. The source device sending the packet after some interval. e.g.
First packet has a time stamp 00:00:00
Second packet has a time stamp 00:00:00.5000000
Third packet has a time stamp 00:00:01
And so on...Means two packet per second.
I want to encode those packets and write to a file.
I am referring the Ffmpeg example from link Muxing.c
While encoding and writing there is no error. But output file has only 2 sec audio duration and speed is also super fast.
The video frames are proper according the settings.
I think the problem is related to calculation of pts, dts and duration of packet.
How should I calculate proper values for pts, dts and duration. Or is this problem related to other thing ?
Code :
void AudioWriter::WriteAudioChunk(IntPtr chunk, int lenght, TimeSpan timestamp)
{
int buffer_size = av_samples_get_buffer_size(NULL, outputStream->tmp_frame->channels, outputStream->tmp_frame->nb_samples, outputStream->AudioStream->codec->sample_fmt, 0);
uint8_t *audioData = reinterpret_cast(static_cast(chunk));
int ret = avcodec_fill_audio_frame(outputStream->tmp_frame,outputStream->Channels, outputStream->AudioStream->codec->sample_fmt, audioData, buffer_size, 1);
if (!ret)
throw gcnew System::IO::IOException("A audio file was not opened yet.");
write_audio_frame(outputStream->FormatContext, outputStream, audioData);
}
static int write_audio_frame(AVFormatContext *oc, AudioWriterData^ ost, uint8_t *audioData)
{
AVCodecContext *c;
AVPacket pkt = { 0 };
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->AudioStream->codec;
AVFrame *frame = ost->tmp_frame;
if (frame)
{
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples, c->sample_rate, c->sample_rate, AV_ROUND_UP);
if (dst_nb_samples != frame->nb_samples)
throw gcnew Exception("dst_nb_samples != frame->nb_samples");
ret = av_frame_make_writable(ost->AudioFrame);
if (ret < 0)
throw gcnew Exception("Unable to make writable.");
ret = swr_convert(ost->swr_ctx, ost->AudioFrame->data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0)
throw gcnew Exception("Unable to convert to destination format.");
frame = ost->AudioFrame;
AVRational timebase = { 1, c->sample_rate };
frame->pts = av_rescale_q(ost->samples_count, timebase, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0)
throw gcnew Exception("Error encoding audio frame.");
if (got_packet)
{
ret = write_frame(oc, &c->time_base, ost->AudioStream, &pkt);
if (ret < 0)
throw gcnew Exception("Audio is not written.");
}
else
throw gcnew Exception("Audio packet encode failed.");
return (ost->AudioFrame || got_packet) ? 0 : 1;
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
return av_interleaved_write_frame(fmt_ctx, pkt);
} -
Android ffmpeg. Linker command failed with exit code 1
8 juin 2017, par Wilsom SandersI’m trying to build ffmpeg static libraries and link them to an android project in order to implement a video player. Goal is to make a player capable of receiving video file from various sources similar to p2p file sharing networks. (target api is 21 level which is 1 level short from supposed official solution with MediaSource)
I managed to compile ffmpeg from this repo (all the code used as-is) but later i got stuck on a linking problem. Whenever I try to compile I get list of ffmpeg methods called in my code and a short eloquent message :
Linker command failed with exit code 1
I have no clue how to pass -v flag to the linker in android studio. Would be great if somebody hinted me that.
I use android studio, build with gradle and cmake.
There’s my files :
build.gradle (Project)// Top-level build file where you can add configuration options common to all sub-projects/modules.
buildscript {
repositories {
jcenter()
}
dependencies {
classpath 'com.android.tools.build:gradle:2.3.1'
// NOTE: Do not place your application dependencies here; they belong
// in the individual module build.gradle files
}
}
allprojects {
repositories {
jcenter()
}
}
task clean(type: Delete) {
delete rootProject.buildDir
}build.gradle (Module)
apply plugin: 'com.android.application'
android {
compileSdkVersion 25
buildToolsVersion "25.0.3"
productFlavors {
x86 {
ndk {
abiFilter "x86"
}
}
arm {
ndk {
abiFilters "armeabi-v7a"
}
}
armv7 {
ndk {
abiFilters "armeabi-v7a"
}
}
}
defaultConfig {
applicationId "com.example.ledo.ndkapplication"
minSdkVersion 22
targetSdkVersion 25
versionCode 1
versionName "1.0"
testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner"
externalNativeBuild {
cmake {
cppFlags "-std=c++11 -frtti -fexceptions"
arguments '-DANDROID_PLATFORM=android-16'
}
}
}
buildTypes {
release {
minifyEnabled false
proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro'
}
}
externalNativeBuild {
cmake {
path "CMakeLists.txt"
}
}
splits {
abi {
enable true
reset()
include 'x86', 'armeabi-v7a'
universalApk true
}
}
}
dependencies {
compile fileTree(dir: 'libs', include: ['*.jar'])
androidTestCompile('com.android.support.test.espresso:espresso-core:2.2.2', {
exclude group: 'com.android.support', module: 'support-annotations'
})
compile 'com.android.support:appcompat-v7:25.3.1'
compile 'com.android.support.constraint:constraint-layout:1.0.2'
compile 'com.android.support:design:25.3.1'
compile 'com.writingminds:FFmpegAndroid:0.3.2'
testCompile 'junit:junit:4.12'
}CMakeLists.txt
# For more information about using CMake with Android Studio, read the
# documentation: https://d.android.com/studio/projects/add-native-code.html
# Sets the minimum version of CMake required to build the native library.
cmake_minimum_required(VERSION 2.8)
# Creates and names a library, sets it as either STATIC
# or SHARED, and provides the relative paths to its source code.
# You can define multiple libraries, and CMake builds them for you.
# Gradle automatically packages shared libraries with your APK.
add_library( # Sets the name of the library.
native-lib
# Sets the library as a shared library.
SHARED
# Provides a relative path to your source file(s).
src/main/cpp/native-lib.cpp
src/main/cpp/NativePlayer.h
src/main/cpp/NativePlayer.cpp)
# Searches for a specified prebuilt library and stores the path as a
# variable. Because CMake includes system libraries in the search path by
# default, you only need to specify the name of the public NDK library
# you want to add. CMake verifies that the library exists before
# completing its build.
find_library( # Sets the name of the path variable.
log-lib
# Specifies the name of the NDK library that
# you want CMake to locate.
log )
find_library(png-lib png)
# Specifies libraries CMake should link to your target library. You
# can link multiple libraries, such as libraries you define in this
# build script, prebuilt third-party libraries, or system libraries.
#avcodec
#avfilter
#avformat
#avutil
#swresample
#swscale
#${ANDROID_ABI}
message(${ANDROID_ABI})
set(FFMPEG_ROOT_DIR src/main/libs/ffmpeg/${ANDROID_ABI})
add_library(avcodec STATIC IMPORTED)
add_library(avformat STATIC IMPORTED)
add_library(avfilter STATIC IMPORTED)
add_library(avutil STATIC IMPORTED)
add_library(swresample STATIC IMPORTED)
add_library(swscale STATIC IMPORTED)
#SET_TARGET_PROPERTIES(avcodec PROPERTIES LINKER_LANGUAGE C)
set_target_properties(avcodec PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libavcodec.a)
#SET_TARGET_PROPERTIES(avfilter PROPERTIES LINKER_LANGUAGE C)
set_target_properties(avfilter PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libavfilter.a)
#SET_TARGET_PROPERTIES(avformat PROPERTIES LINKER_LANGUAGE C)
set_target_properties(avformat PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libavformat.a)
#SET_TARGET_PROPERTIES(avutil PROPERTIES LINKER_LANGUAGE C)
set_target_properties(avutil PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libavutil.a)
#SET_TARGET_PROPERTIES(swresample PROPERTIES LINKER_LANGUAGE C)
set_target_properties(swresample PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libswresample.a)
#SET_TARGET_PROPERTIES(swscale PROPERTIES LINKER_LANGUAGE C)
set_target_properties(swscale PROPERTIES IMPORTED_LOCATION ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib/libswscale.a)
include_directories( ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/include )
include_directories( ${CMAKE_CURRENT_SOURCE_DIR}/${FFMPEG_ROOT_DIR}/lib )
target_link_libraries( # Specifies the target library.
native-lib
# Links the target library to the log library
# included in the NDK.
${log-lib}
avcodec
avformat
#avfilter
#swresample
#swscale
#avutil
GLESv2)I have .a files in following locations :
%PROJECT_DIR%/app/src/libs/ffmpeg/armeabi-v7a
%PROJECT_DIR%/app/src/libs/ffmpeg/armeabi-v7a-neon
%PROJECT_DIR%/app/src/libs/ffmpeg/x86It doesn’t look to me that linker misses files themselves. (I get different out put if I misplace them.)
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A ffmpeg comman canwork in cmd but not in Python using subprocess.call() or os.system()
6 juin 2018, par StarryskyI wanna transfer a .mp3 to .wav. This is my command :
ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wavIt worked well in cmd
C:\Users\starrysky\Documents\GitHub\bing_pic\html>ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav
ffmpeg version N-86482-gbc40674 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 7.1.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
libavutil 55. 66.100 / 55. 66.100
libavcodec 57. 99.100 / 57. 99.100
libavformat 57. 73.100 / 57. 73.100
libavdevice 57. 7.100 / 57. 7.100
libavfilter 6. 92.100 / 6. 92.100
libswscale 4. 7.101 / 4. 7.101
libswresample 2. 8.100 / 2. 8.100
libpostproc 54. 6.100 / 54. 6.100
Input #0, mp3, from 'a.mp3':
Metadata:
encoder : Lavf54.6.100
Duration: 00:00:01.87, start: 0.000000, bitrate: 8 kb/s
Stream #0:0: Audio: mp3, 8000 Hz, mono, s16p, 8 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'a.wav':
Metadata:
ISFT : Lavf57.73.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
Metadata:
encoder : Lavc57.99.100 pcm_s16le
size= 59kB time=00:00:01.87 bitrate= 256.3kbits/s speed= 187x
video:0kB audio:58kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.130208%but when I moved it into my python program, something strange happened.
>>> C:\Users\starrysky\Documents\GitHub\bing_pic\html\
'ffmpeg' �����ڲ����ⲿ���Ҳ���ǿ����еij���
���������ļ���
1 Command 'ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav' returned non-zero exit status 1.
文件错误啊，亲
[WinError 2] 系统找不到指定的文件。: 'a.wav'This is part of my python code :
@bot.register(wife, RECORDING)
def translate_sound(msg):
msg.get_file(save_path='a.mp3')
path = os.path.abspath('.')+'\\'
print(path)
try:
subprocess.check_call('ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav', shell=True)
# ''
except Exception as e:
print(1, e)
wav_to_text('a.wav')
try:
os.remove('a.wav')
except Exception as e:
print(e)# 调用百度语音识别API
def get_token():
URL = 'http://openapi.baidu.com/oauth/2.0/token'
_params = urllib.parse.urlencode({'grant_type': b'client_credentials',
'client_id': b''
'client_secret': b''})
_res = urllib.request.Request(URL, _params.encode())
_response = urllib.request.urlopen(_res)
_data = _response.read()
_data = json.loads(_data)
return _data['access_token']
def wav_to_text(wav_file):
try:
wav_file = open(wav_file, 'rb')
except IOError:
print('文件错误啊，亲')
return
wav_file = wave.open(wav_file)
n_frames = wav_file.getnframes()
print('n_frames ', n_frames)
frame_rate = wav_file.getframerate()
print("frame_rate ", frame_rate)
if n_frames == 1 or frame_rate not in (8000, 16000):
print('不符合格式')
return
audio = wav_file.readframes(n_frames)
seconds = n_frames/frame_rate+1
minute = int(seconds/60 + 1)
for i in range(0, minute):
sub_audio = audio[i*60*frame_rate:(i+1)*60*frame_rate]
base_data = base64.b64encode(sub_audio)
data = {"format": "wav",
"token": get_token(),
"len": len(sub_audio),
"rate": frame_rate,
"speech": base_data.decode(),
"cuid": "B8-AC-6F-2D-7A-94",
"channel": 1}
data = json.dumps(data)
res = urllib.request.Request('http://vop.baidu.com/server_api',
data.encode(),
{'content-type': 'application/json'})
response = urllib.request.urlopen(res)
res_data = json.loads(response.read())
try:
print(res_data['result'][0])
except Exception as e:
print(e)What happened ?
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