Recherche avancée
Médias (3)
-
Elephants Dream - Cover of the soundtrack
17 octobre 2011, par
Mis à jour : Octobre 2011
Langue : English
Type : Image
-
Valkaama DVD Label
4 octobre 2011, par
Mis à jour : Février 2013
Langue : English
Type : Image
-
Publier une image simplement
13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (58)
-
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 (...) -
La sauvegarde automatique de canaux SPIP
1er avril 2010, parDans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (9694)
-
Revision d205335060 : [svc] Finalize spatial svc first pass rate control 1. Save stats for each
19 mars 2014, par Minghai ShangChanged Paths :
Modify /examples/vp9_spatial_scalable_encoder.c
Modify /test/svc_test.cc
Modify /vp9/encoder/vp9_firstpass.c
Modify /vp9/encoder/vp9_firstpass.h
Modify /vp9/encoder/vp9_onyx_if.c
Modify /vp9/encoder/vp9_onyx_int.h
Modify /vp9/encoder/vp9_svc_layercontext.h
Modify /vpx/src/svc_encodeframe.c
Modify /vpx/vpx_encoder.h
[svc] Finalize spatial svc first pass rate control1. Save stats for each spatial layer
2. Add frame buffer management for svc first pass rc
3. Set default spatial layer to 1
4. Flush encoder at the end of stream in test app
This only supports spatial svc.
Change-Id : Ia89cfa87bb6394e6c0405b921d86c426d0a0c9ae -
How to debug ffmpeg reliability for long running rtsp streams
13 septembre 2022, par MarkI have a long running ffmpeg background process that "watches" an rtsp stream and takes snapshots every 7 minutes.


It's being run like this


C:\Windows\System32\cmd.exe /c C:\ffmpeg\bin\ffmpeg.exe -nostdin -rtsp_transport tcp -y -timeout 5000000 -i rtsp://someurl -q:v 1 -an -vf fps=0.002381,scale="1280:720" -strftime 1 -f image2 C:\somelocalfolder\%Y-%m-%d_%H-%M-%S.jpg > c:\ffmpeglog.txt 2>&1


This process runs for days but intermittently, for hours at a time, seems to miss taking snapshots, until eventually it starts to take them again - then fail again, etc. The logs at info level are not helpful. I checked the stream during times when it was not taking snapshots and the stream was up. What's happening here ? How can I debug this ?


Below is an image of succesfull snapshots per hour. There should always be between 8 and 9.



Logs look like this


ffmpeg version 2022-03-31-git-e301a24fa1-full_build-www.gyan.dev Copyright (c) 2000-2022 the FFmpeg developers
 built with gcc 11.2.0 (Rev7, Built by MSYS2 project)
 configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
 libavutil 57. 24.101 / 57. 24.101
 libavcodec 59. 25.100 / 59. 25.100
 libavformat 59. 20.101 / 59. 20.101
 libavdevice 59. 6.100 / 59. 6.100
 libavfilter 8. 29.100 / 8. 29.100
 libswscale 6. 6.100 / 6. 6.100
 libswresample 4. 6.100 / 4. 6.100
 libpostproc 56. 5.100 / 56. 5.100
Input #0, rtsp, from 'rtsp://somerul':
 Metadata:
 title : HIK Media Server V4.21.005
 comment : HIK Media Server Session Description : standard
 Duration: N/A, start: 0.033000, bitrate: N/A
 Stream #0:0: Video: h264 (High), yuv420p(progressive), 704x576, 30 tbr, 90k tbn
Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2c2e0c0] deprecated pixel format used, make sure you did set range correctly
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2c67c40] deprecated pixel format used, make sure you did set range correctly
[swscaler @ 000002a1c2c20680] [swscaler @ 000002a1c2cc6700] deprecated pixel format used, make sure you did set range correctly
Output #0, image2, to 'C:\somelocalfolder\Temp\stream_2\StreamedImages\%Y-%m-%d_%H-%M-%S.jpg':
 Metadata:
 title : HIK Media Server V4.21.005
 comment : HIK Media Server Session Description : standard
 encoder : Lavf59.20.101
 Stream #0:0: Video: mjpeg, yuvj420p(pc, progressive), 1280x720, q=2-31, 200 kb/s, 0.0024 fps, 0.0024 tbn
 Metadata:
 encoder : Lavc59.25.100 mjpeg
 Side data:
 cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: N/A
frame= 1 fps=0.0 q=0.0 size=N/A time=00:00:00.00 bitrate=N/A speed= 0x 


Update
I got some trace logs. The ffmpeg seems to fail silently at some point and stop taking snapshots.


After about 3 million log lines (which is really only a couple of hours in my case) I get the following


rtsp://192.168.15.195:554/streaming/channels/904: Unknown error


But ffmpeg silently continues. Here is a bit more of the log


[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074443040, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.95x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[h264 @ 00000248e7d59880] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[rtsp @ 00000248e765cf00] ret=-138 c=24 [$]
rtsp://192.168.15.195:554/streaming/channels/904: Unknown error
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074446100, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.95x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=696
[rtsp @ 00000248e765cf00] Sending:
GET_PARAMETER rtsp://192.168.15.195:554/streaming/channels/904 RTSP/1.0

CSeq: 402

User-Agent: Lavf59.20.101

Session: 931848797

Authorization: Digest username="******", realm="709382dda4ccb674edf093d3", nonce="13fca62fc", uri="rtsp://192.168.15.195:554/streaming/channels/904", response="74341df9611f0ac3dc247b402424735b", algorithm="MD5"



--
[NULL @ 00000248e7662640] nal_unit_type: 7(SPS), nal_ref_idc: 3
[NULL @ 00000248e7662640] nal_unit_type: 8(PPS), nal_ref_idc: 3
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=756
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074449070, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Read frame with in pts 1074449070, out pts 28
[Parsed_fps_0 @ 00000248e7d50e40] Dropping frame with pts 28
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.949x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
frame= 28 fps=0.0 q=1.0 size=N/A time=03:08:59.77 bitrate=N/A speed=0.949x 
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1228
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[NULL @ 00000248e7662640] reference count 1 overflow
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=804
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=1352
[NULL @ 00000248e7662640] illegal memory management control operation 11
[rtsp @ 00000248e765cf00] tcp_read_packet:
[rtsp @ 00000248e765cf00] ret=1 c=24 [$]
[rtsp @ 00000248e765cf00] id=0 len=836


Basically it appears an issue of ffmpeg silently failing. If it crashed, my software could detect it and I could rerun it, but if it fails silently like this, I need another solution.


-
ffmpeg : libavformat/libswresample to transcode and resample at same time
21 février 2024, par whatdoidoI want to transcode and down/re-sample the audio for output using ffmpeg's libav*/libswresample - I am using ffmpeg's (4.x) transcode_aac.c and resample_audio.c as reference - but the code produces audio with glitches that is clearly not what ffmpeg itself would produce (ie ffmpeg -i foo.wav -ar 22050 foo.m4a)


Based on the ffmpeg examples, to resample audio it appears that I need to set the output AVAudioContext and SwrContext sample_rate to what I desire and ensure the swr_convert() is provided with the correct number of output samples based av_rescale_rnd( swr_delay(), ...) once I have an decoded input audio. I've taken care to ensure all the relevant calculations of samples for output are taken into account in the merged code (below) :


- 

- open_output_file() - AVCodecContext.sample_rate (avctx variable) set to our target (down sampled) sample_rate
- read_decode_convert_and_store() is where the work happens : input audio is decoded to an AVFrame and this input frame is converted before being encoded.

- 

- init_converted_samples() and av_samples_alloc() uses the input frame's nb_samples
- ADDED : calc the number of output samples via av_rescale_rnd() and swr_delay()
- UPDATED : convert_samples() and swr_convert() uses the input frame's samples and our calculated output samples as parameters














However the resulting audio file is produced with audio glitches. Does the community know of any references for how transcode AND resample should be done or what is missing in this example ?


/* compile and run:
 gcc -I/usr/include/ffmpeg transcode-swr-aac.c -lavformat -lavutil -lavcodec -lswresample -lm
 ./a.out foo.wav foo.m4a
 */

/*
 * Copyright (c) 2013-2018 Andreas Unterweger
 * 
 * This file is part of FFmpeg. 
 ... ...
 * 
 * @example transcode_aac.c 
 * Convert an input audio file to AAC in an MP4 container using FFmpeg. 
 * Formats other than MP4 are supported based on the output file extension. 
 * @author Andreas Unterweger (xxxx@xxxxx.com)
 */ 
 #include 
 

 #include "libavformat/avformat.h"
 #include "libavformat/avio.h"
 
 #include "libavcodec/avcodec.h"
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/frame.h"
 #include "libavutil/opt.h"
 
 #include "libswresample/swresample.h"
 
 #define OUTPUT_BIT_RATE 128000
 #define OUTPUT_CHANNELS 2
 
 static int open_input_file(const char *filename,
 AVFormatContext **input_format_context,
 AVCodecContext **input_codec_context)
 {
 AVCodecContext *avctx;
 const AVCodec *input_codec;
 const AVStream *stream;
 int error;
 
 if ((error = avformat_open_input(input_format_context, filename, NULL,
 NULL)) < 0) {
 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
 filename, av_err2str(error));
 *input_format_context = NULL;
 return error;
 }
 

 if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
 fprintf(stderr, "Could not open find stream info (error '%s')\n",
 av_err2str(error));
 avformat_close_input(input_format_context);
 return error;
 }
 
 if ((*input_format_context)->nb_streams != 1) {
 fprintf(stderr, "Expected one audio input stream, but found %d\n",
 (*input_format_context)->nb_streams);
 avformat_close_input(input_format_context);
 return AVERROR_EXIT;
 }
 
 stream = (*input_format_context)->streams[0];
 
 if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
 fprintf(stderr, "Could not find input codec\n");
 avformat_close_input(input_format_context);
 return AVERROR_EXIT;
 }
 
 avctx = avcodec_alloc_context3(input_codec);
 if (!avctx) {
 fprintf(stderr, "Could not allocate a decoding context\n");
 avformat_close_input(input_format_context);
 return AVERROR(ENOMEM);
 }
 
 /* Initialize the stream parameters with demuxer information. */
 error = avcodec_parameters_to_context(avctx, stream->codecpar);
 if (error < 0) {
 avformat_close_input(input_format_context);
 avcodec_free_context(&avctx);
 return error;
 }
 
 /* Open the decoder for the audio stream to use it later. */
 if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
 fprintf(stderr, "Could not open input codec (error '%s')\n",
 av_err2str(error));
 avcodec_free_context(&avctx);
 avformat_close_input(input_format_context);
 return error;
 }
 
 /* Set the packet timebase for the decoder. */
 avctx->pkt_timebase = stream->time_base;
 
 /* Save the decoder context for easier access later. */
 *input_codec_context = avctx;
 
 return 0;
 }
 
 static int open_output_file(const char *filename,
 AVCodecContext *input_codec_context,
 AVFormatContext **output_format_context,
 AVCodecContext **output_codec_context)
 {
 AVCodecContext *avctx = NULL;
 AVIOContext *output_io_context = NULL;
 AVStream *stream = NULL;
 const AVCodec *output_codec = NULL;
 int error;
 

 if ((error = avio_open(&output_io_context, filename,
 AVIO_FLAG_WRITE)) < 0) {
 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
 filename, av_err2str(error));
 return error;
 }
 

 if (!(*output_format_context = avformat_alloc_context())) {
 fprintf(stderr, "Could not allocate output format context\n");
 return AVERROR(ENOMEM);
 }
 

 (*output_format_context)->pb = output_io_context;
 

 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
 NULL))) {
 fprintf(stderr, "Could not find output file format\n");
 goto cleanup;
 }
 
 if (!((*output_format_context)->url = av_strdup(filename))) {
 fprintf(stderr, "Could not allocate url.\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 

 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
 fprintf(stderr, "Could not find an AAC encoder.\n");
 goto cleanup;
 }
 
 /* Create a new audio stream in the output file container. */
 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
 fprintf(stderr, "Could not create new stream\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 
 avctx = avcodec_alloc_context3(output_codec);
 if (!avctx) {
 fprintf(stderr, "Could not allocate an encoding context\n");
 error = AVERROR(ENOMEM);
 goto cleanup;
 }
 
 /* Set the basic encoder parameters.
 * SET OUR DESIRED output sample_rate here
 */
 avctx->channels = OUTPUT_CHANNELS;
 avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
 // avctx->sample_rate = input_codec_context->sample_rate;
 avctx->sample_rate = 22050;
 avctx->sample_fmt = output_codec->sample_fmts[0];
 avctx->bit_rate = OUTPUT_BIT_RATE;
 
 avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
 
 /* Set the sample rate for the container. */
 stream->time_base.den = avctx->sample_rate;
 stream->time_base.num = 1;
 
 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
 avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
 fprintf(stderr, "Could not open output codec (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 error = avcodec_parameters_from_context(stream->codecpar, avctx);
 if (error < 0) {
 fprintf(stderr, "Could not initialize stream parameters\n");
 goto cleanup;
 }
 
 /* Save the encoder context for easier access later. */
 *output_codec_context = avctx;
 
 return 0;
 
 cleanup:
 avcodec_free_context(&avctx);
 avio_closep(&(*output_format_context)->pb);
 avformat_free_context(*output_format_context);
 *output_format_context = NULL;
 return error < 0 ? error : AVERROR_EXIT;
 }
 
 /**
 * Initialize one data packet for reading or writing.
 */
 static int init_packet(AVPacket **packet)
 {
 if (!(*packet = av_packet_alloc())) {
 fprintf(stderr, "Could not allocate packet\n");
 return AVERROR(ENOMEM);
 }
 return 0;
 }
 
 static int init_input_frame(AVFrame **frame)
 {
 if (!(*frame = av_frame_alloc())) {
 fprintf(stderr, "Could not allocate input frame\n");
 return AVERROR(ENOMEM);
 }
 return 0;
 }
 
 static int init_resampler(AVCodecContext *input_codec_context,
 AVCodecContext *output_codec_context,
 SwrContext **resample_context)
 {
 int error;

 /**
 * create the resample, including ref to the desired output sample rate
 */
 *resample_context = swr_alloc_set_opts(NULL,
 av_get_default_channel_layout(output_codec_context->channels),
 output_codec_context->sample_fmt,
 output_codec_context->sample_rate,
 av_get_default_channel_layout(input_codec_context->channels),
 input_codec_context->sample_fmt,
 input_codec_context->sample_rate,
 0, NULL);
 if (!*resample_context < 0) {
 fprintf(stderr, "Could not allocate resample context\n");
 return AVERROR(ENOMEM);
 }
 
 if ((error = swr_init(*resample_context)) < 0) {
 fprintf(stderr, "Could not open resample context\n");
 swr_free(resample_context);
 return error;
 }
 return 0;
 }
 
 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
 if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
 output_codec_context->channels, 1))) {
 fprintf(stderr, "Could not allocate FIFO\n");
 return AVERROR(ENOMEM);
 }
 return 0;
 }
 
 static int write_output_file_header(AVFormatContext *output_format_context)
 {
 int error;
 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
 fprintf(stderr, "Could not write output file header (error '%s')\n",
 av_err2str(error));
 return error;
 }
 return 0;
 }
 
 static int decode_audio_frame(AVFrame *frame,
 AVFormatContext *input_format_context,
 AVCodecContext *input_codec_context,
 int *data_present, int *finished)
 {
 AVPacket *input_packet;
 int error;
 
 error = init_packet(&input_packet);
 if (error < 0)
 return error;
 
 *data_present = 0;
 *finished = 0;

 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
 if (error == AVERROR_EOF)
 *finished = 1;
 else {
 fprintf(stderr, "Could not read frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 }
 
 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 error = avcodec_receive_frame(input_codec_context, frame);
 if (error == AVERROR(EAGAIN)) {
 error = 0;
 goto cleanup;
 } else if (error == AVERROR_EOF) {
 *finished = 1;
 error = 0;
 goto cleanup;
 } else if (error < 0) {
 fprintf(stderr, "Could not decode frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 } else {
 *data_present = 1;
 goto cleanup;
 }
 
 cleanup:
 av_packet_free(&input_packet);
 return error;
 }
 
 static int init_converted_samples(uint8_t ***converted_input_samples,
 AVCodecContext *output_codec_context,
 int frame_size)
 {
 int error;
 
 if (!(*converted_input_samples = calloc(output_codec_context->channels,
 sizeof(**converted_input_samples)))) {
 fprintf(stderr, "Could not allocate converted input sample pointers\n");
 return AVERROR(ENOMEM);
 }
 

 if ((error = av_samples_alloc(*converted_input_samples, NULL,
 output_codec_context->channels,
 frame_size,
 output_codec_context->sample_fmt, 0)) < 0) {
 fprintf(stderr,
 "Could not allocate converted input samples (error '%s')\n",
 av_err2str(error));
 av_freep(&(*converted_input_samples)[0]);
 free(*converted_input_samples);
 return error;
 }
 return 0;
 }
 
 static int convert_samples(const uint8_t **input_data, const int input_nb_samples,
 uint8_t **converted_data, const int output_nb_samples,
 SwrContext *resample_context)
 {
 int error;
 
 if ((error = swr_convert(resample_context,
 converted_data, output_nb_samples,
 input_data , input_nb_samples)) < 0) {
 fprintf(stderr, "Could not convert input samples (error '%s')\n",
 av_err2str(error));
 return error;
 }
 
 return 0;
 }
 
 static int add_samples_to_fifo(AVAudioFifo *fifo,
 uint8_t **converted_input_samples,
 const int frame_size)
 {
 int error;
 
 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
 fprintf(stderr, "Could not reallocate FIFO\n");
 return error;
 }
 
 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
 frame_size) < frame_size) {
 fprintf(stderr, "Could not write data to FIFO\n");
 return AVERROR_EXIT;
 }
 return 0;
 }
 
 static int read_decode_convert_and_store(AVAudioFifo *fifo,
 AVFormatContext *input_format_context,
 AVCodecContext *input_codec_context,
 AVCodecContext *output_codec_context,
 SwrContext *resampler_context,
 int *finished)
 {
 AVFrame *input_frame = NULL;
 uint8_t **converted_input_samples = NULL;
 int data_present;
 int ret = AVERROR_EXIT;
 

 if (init_input_frame(&input_frame))
 goto cleanup;

 if (decode_audio_frame(input_frame, input_format_context,
 input_codec_context, &data_present, finished))
 goto cleanup;

 if (*finished) {
 ret = 0;
 goto cleanup;
 }

 if (data_present) {
 /* Initialize the temporary storage for the converted input samples. */
 if (init_converted_samples(&converted_input_samples, output_codec_context,
 input_frame->nb_samples))
 goto cleanup;
 
 /* figure out how many samples are required for target sample_rate incl
 * any items left in the swr buffer
 */ 
 int output_nb_samples = av_rescale_rnd(
 swr_get_delay(resampler_context, input_codec_context->sample_rate) + input_frame->nb_samples,
 output_codec_context->sample_rate, 
 input_codec_context->sample_rate,
 AV_ROUND_UP);
 
 /* ignore, just to ensure we've got enough buffer alloc'd for conversion buffer */
 av_assert1(input_frame->nb_samples > output_nb_samples);
 
 /* Convert the input samples to the desired output sample format, via swr_convert().
 */
 if (convert_samples((const uint8_t**)input_frame->extended_data, input_frame->nb_samples,
 converted_input_samples, output_nb_samples,
 resampler_context))
 goto cleanup;
 
 /* Add the converted input samples to the FIFO buffer for later processing. */
 if (add_samples_to_fifo(fifo, converted_input_samples,
 output_nb_samples))
 goto cleanup;
 ret = 0;
 }
 ret = 0;
 
 cleanup:
 if (converted_input_samples) {
 av_freep(&converted_input_samples[0]);
 free(converted_input_samples);
 }
 av_frame_free(&input_frame);
 
 return ret;
 }
 
 static int init_output_frame(AVFrame **frame,
 AVCodecContext *output_codec_context,
 int frame_size)
 {
 int error;
 
 if (!(*frame = av_frame_alloc())) {
 fprintf(stderr, "Could not allocate output frame\n");
 return AVERROR_EXIT;
 }
 
 /* Set the frame's parameters, especially its size and format.
 * av_frame_get_buffer needs this to allocate memory for the
 * audio samples of the frame.
 * Default channel layouts based on the number of channels
 * are assumed for simplicity. */
 (*frame)->nb_samples = frame_size;
 (*frame)->channel_layout = output_codec_context->channel_layout;
 (*frame)->format = output_codec_context->sample_fmt;
 (*frame)->sample_rate = output_codec_context->sample_rate;
 
 /* Allocate the samples of the created frame. This call will make
 * sure that the audio frame can hold as many samples as specified. */
 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
 av_err2str(error));
 av_frame_free(frame);
 return error;
 }
 
 return 0;
 }
 
 /* Global timestamp for the audio frames. */
 static int64_t pts = 0;
 
 /**
 * Encode one frame worth of audio to the output file.
 */
 static int encode_audio_frame(AVFrame *frame,
 AVFormatContext *output_format_context,
 AVCodecContext *output_codec_context,
 int *data_present)
 {
 AVPacket *output_packet;
 int error;
 
 error = init_packet(&output_packet);
 if (error < 0)
 return error;
 
 /* Set a timestamp based on the sample rate for the container. */
 if (frame) {
 frame->pts = pts;
 pts += frame->nb_samples;
 }
 
 *data_present = 0;
 error = avcodec_send_frame(output_codec_context, frame);
 if (error < 0 && error != AVERROR_EOF) {
 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 

 error = avcodec_receive_packet(output_codec_context, output_packet);
 if (error == AVERROR(EAGAIN)) {
 error = 0;
 goto cleanup;
 } else if (error == AVERROR_EOF) {
 error = 0;
 goto cleanup;
 } else if (error < 0) {
 fprintf(stderr, "Could not encode frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 } else {
 *data_present = 1;
 }
 
 /* Write one audio frame from the temporary packet to the output file. */
 if (*data_present &&
 (error = av_write_frame(output_format_context, output_packet)) < 0) {
 fprintf(stderr, "Could not write frame (error '%s')\n",
 av_err2str(error));
 goto cleanup;
 }
 
 cleanup:
 av_packet_free(&output_packet);
 return error;
 }
 
 /**
 * Load one audio frame from the FIFO buffer, encode and write it to the
 * output file.
 */
 static int load_encode_and_write(AVAudioFifo *fifo,
 AVFormatContext *output_format_context,
 AVCodecContext *output_codec_context)
 {
 AVFrame *output_frame;
 /* Use the maximum number of possible samples per frame.
 * If there is less than the maximum possible frame size in the FIFO
 * buffer use this number. Otherwise, use the maximum possible frame size. */
 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
 output_codec_context->frame_size);
 int data_written;
 
 if (init_output_frame(&output_frame, output_codec_context, frame_size))
 return AVERROR_EXIT;
 
 /* Read as many samples from the FIFO buffer as required to fill the frame.
 * The samples are stored in the frame temporarily. */
 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
 fprintf(stderr, "Could not read data from FIFO\n");
 av_frame_free(&output_frame);
 return AVERROR_EXIT;
 }
 
 /* Encode one frame worth of audio samples. */
 if (encode_audio_frame(output_frame, output_format_context,
 output_codec_context, &data_written)) {
 av_frame_free(&output_frame);
 return AVERROR_EXIT;
 }
 av_frame_free(&output_frame);
 return 0;
 }
 
 /**
 * Write the trailer of the output file container.
 */
 static int write_output_file_trailer(AVFormatContext *output_format_context)
 {
 int error;
 if ((error = av_write_trailer(output_format_context)) < 0) {
 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
 av_err2str(error));
 return error;
 }
 return 0;
 }
 
 int main(int argc, char **argv)
 {
 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
 SwrContext *resample_context = NULL;
 AVAudioFifo *fifo = NULL;
 int ret = AVERROR_EXIT;
 
 if (argc != 3) {
 fprintf(stderr, "Usage: %s <input file="file" /> <output file="file">\n", argv[0]);
 exit(1);
 }
 

 if (open_input_file(argv[1], &input_format_context,
 &input_codec_context))
 goto cleanup;

 if (open_output_file(argv[2], input_codec_context,
 &output_format_context, &output_codec_context))
 goto cleanup;

 if (init_resampler(input_codec_context, output_codec_context,
 &resample_context))
 goto cleanup;

 if (init_fifo(&fifo, output_codec_context))
 goto cleanup;

 if (write_output_file_header(output_format_context))
 goto cleanup;
 
 while (1) {
 /* Use the encoder's desired frame size for processing. */
 const int output_frame_size = output_codec_context->frame_size;
 int finished = 0;
 
 while (av_audio_fifo_size(fifo) < output_frame_size) {
 /* Decode one frame worth of audio samples, convert it to the
 * output sample format and put it into the FIFO buffer. */
 if (read_decode_convert_and_store(fifo, input_format_context,
 input_codec_context,
 output_codec_context,
 resample_context, &finished))
 goto cleanup;
 
 if (finished)
 break;
 }
 
 while (av_audio_fifo_size(fifo) >= output_frame_size ||
 (finished && av_audio_fifo_size(fifo) > 0))
 if (load_encode_and_write(fifo, output_format_context,
 output_codec_context))
 goto cleanup;
 
 if (finished) {
 int data_written;
 do {
 if (encode_audio_frame(NULL, output_format_context,
 output_codec_context, &data_written))
 goto cleanup;
 } while (data_written);
 break;
 }
 }
 
 if (write_output_file_trailer(output_format_context))
 goto cleanup;
 ret = 0;
 
 cleanup:
 if (fifo)
 av_audio_fifo_free(fifo);
 swr_free(&resample_context);
 if (output_codec_context)
 avcodec_free_context(&output_codec_context);
 if (output_format_context) {
 avio_closep(&output_format_context->pb);
 avformat_free_context(output_format_context);
 }
 if (input_codec_context)
 avcodec_free_context(&input_codec_context);
 if (input_format_context)
 avformat_close_input(&input_format_context);
 
 return ret;
 }
</output>

Syndiquer tout le site
