As it stands, my project correctly uses libavcodec to decode a video, where each frame is manipulated (it doesn't matter how) and output to a new video. I've cobbled this together from examples found online, and it works. The result is a perfect .mp4 of the manipulated frames, minus the audio.
My problem is, when I try to add an audio stream to the output container, I get a crash in mux.c that I can't explain. It's in static int compute_muxer_pkt_fields(AVFormatContext *s, AVStream *st, AVPacket *pkt). Where st->internal->priv_pts->val = pkt->dts; is attempted, priv_pts is nullptr.
I don't recall the version number, but this is from a November 4, 2020 ffmpeg build from git.
My MediaContentMgr is much bigger than what I have here. I'm stripping out everything to do with the frame manipulation, so if I'm missing anything, please let me know and I'll edit.
The code that, when added, triggers the nullptr exception, is called out inline
The .h:
#ifndef _API_EXAMPLE_H
#define _API_EXAMPLE_H
#include <glad/glad.h>
#include <GLFW/glfw3.h>
#include "glm/glm.hpp"
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libavutil/opt.h>
#include <libswscale/swscale.h>
}
#include "shader_s.h"
class MediaContainerMgr {
public:
MediaContainerMgr(const std::string& infile, const std::string& vert, const std::string& frag,
const glm::vec3* extents);
~MediaContainerMgr();
void render();
bool recording() { return m_recording; }
// Major thanks to "shi-yan" who helped make this possible:
// https://github.com/shi-yan/videosamples/blob/master/libavmp4encoding/main.cpp
bool init_video_output(const std::string& video_file_name, unsigned int width, unsigned int height);
bool output_video_frame(uint8_t* buf);
bool finalize_output();
private:
AVFormatContext* m_format_context;
AVCodec* m_video_codec;
AVCodec* m_audio_codec;
AVCodecParameters* m_video_codec_parameters;
AVCodecParameters* m_audio_codec_parameters;
AVCodecContext* m_codec_context;
AVFrame* m_frame;
AVPacket* m_packet;
uint32_t m_video_stream_index;
uint32_t m_audio_stream_index;
void init_rendering(const glm::vec3* extents);
int decode_packet();
// For writing the output video:
void free_output_assets();
bool m_recording;
AVOutputFormat* m_output_format;
AVFormatContext* m_output_format_context;
AVCodec* m_output_video_codec;
AVCodecContext* m_output_video_codec_context;
AVFrame* m_output_video_frame;
SwsContext* m_output_scale_context;
AVStream* m_output_video_stream;
AVCodec* m_output_audio_codec;
AVStream* m_output_audio_stream;
AVCodecContext* m_output_audio_codec_context;
};
#endif
And, the hellish .cpp:
#include <stdio.h>
#include <stdarg.h>
#include <stdlib.h>
#include <string.h>
#include <inttypes.h>
#include "media_container_manager.h"
MediaContainerMgr::MediaContainerMgr(const std::string& infile, const std::string& vert, const std::string& frag,
const glm::vec3* extents) :
m_video_stream_index(-1),
m_audio_stream_index(-1),
m_recording(false),
m_output_format(nullptr),
m_output_format_context(nullptr),
m_output_video_codec(nullptr),
m_output_video_codec_context(nullptr),
m_output_video_frame(nullptr),
m_output_scale_context(nullptr),
m_output_video_stream(nullptr)
{
// AVFormatContext holds header info from the format specified in the container:
m_format_context = avformat_alloc_context();
if (!m_format_context) {
throw "ERROR could not allocate memory for Format Context";
}
// open the file and read its header. Codecs are not opened here.
if (avformat_open_input(&m_format_context, infile.c_str(), NULL, NULL) != 0) {
throw "ERROR could not open input file for reading";
}
printf("format %s, duration %lldus, bit_rate %lld\n", m_format_context->iformat->name, m_format_context->duration, m_format_context->bit_rate);
//read avPackets (?) from the avFormat (?) to get stream info. This populates format_context->streams.
if (avformat_find_stream_info(m_format_context, NULL) < 0) {
throw "ERROR could not get stream info";
}
for (unsigned int i = 0; i < m_format_context->nb_streams; i++) {
AVCodecParameters* local_codec_parameters = NULL;
local_codec_parameters = m_format_context->streams[i]->codecpar;
printf("AVStream->time base before open coded %d/%d\n", m_format_context->streams[i]->time_base.num, m_format_context->streams[i]->time_base.den);
printf("AVStream->r_frame_rate before open coded %d/%d\n", m_format_context->streams[i]->r_frame_rate.num, m_format_context->streams[i]->r_frame_rate.den);
printf("AVStream->start_time %" PRId64 "\n", m_format_context->streams[i]->start_time);
printf("AVStream->duration %" PRId64 "\n", m_format_context->streams[i]->duration);
printf("duration(s): %lf\n", (float)m_format_context->streams[i]->duration / m_format_context->streams[i]->time_base.den * m_format_context->streams[i]->time_base.num);
AVCodec* local_codec = NULL;
local_codec = avcodec_find_decoder(local_codec_parameters->codec_id);
if (local_codec == NULL) {
throw "ERROR unsupported codec!";
}
if (local_codec_parameters->codec_type == AVMEDIA_TYPE_VIDEO) {
if (m_video_stream_index == -1) {
m_video_stream_index = i;
m_video_codec = local_codec;
m_video_codec_parameters = local_codec_parameters;
}
m_height = local_codec_parameters->height;
m_width = local_codec_parameters->width;
printf("Video Codec: resolution %dx%d\n", m_width, m_height);
}
else if (local_codec_parameters->codec_type == AVMEDIA_TYPE_AUDIO) {
if (m_audio_stream_index == -1) {
m_audio_stream_index = i;
m_audio_codec = local_codec;
m_audio_codec_parameters = local_codec_parameters;
}
printf("Audio Codec: %d channels, sample rate %d\n", local_codec_parameters->channels, local_codec_parameters->sample_rate);
}
printf("\tCodec %s ID %d bit_rate %lld\n", local_codec->name, local_codec->id, local_codec_parameters->bit_rate);
}
m_codec_context = avcodec_alloc_context3(m_video_codec);
if (!m_codec_context) {
throw "ERROR failed to allocate memory for AVCodecContext";
}
if (avcodec_parameters_to_context(m_codec_context, m_video_codec_parameters) < 0) {
throw "ERROR failed to copy codec params to codec context";
}
if (avcodec_open2(m_codec_context, m_video_codec, NULL) < 0) {
throw "ERROR avcodec_open2 failed to open codec";
}
m_frame = av_frame_alloc();
if (!m_frame) {
throw "ERROR failed to allocate AVFrame memory";
}
m_packet = av_packet_alloc();
if (!m_packet) {
throw "ERROR failed to allocate AVPacket memory";
}
}
MediaContainerMgr::~MediaContainerMgr() {
avformat_close_input(&m_format_context);
av_packet_free(&m_packet);
av_frame_free(&m_frame);
avcodec_free_context(&m_codec_context);
glDeleteVertexArrays(1, &m_VAO);
glDeleteBuffers(1, &m_VBO);
}
bool MediaContainerMgr::advance_frame() {
while (true) {
if (av_read_frame(m_format_context, m_packet) < 0) {
// Do we actually need to unref the packet if it failed?
av_packet_unref(m_packet);
continue;
//return false;
}
else {
if (m_packet->stream_index == m_video_stream_index) {
//printf("AVPacket->pts %" PRId64 "\n", m_packet->pts);
int response = decode_packet();
av_packet_unref(m_packet);
if (response != 0) {
continue;
//return false;
}
return true;
}
else {
printf("m_packet->stream_index: %d\n", m_packet->stream_index);
printf(" m_packet->pts: %lld\n", m_packet->pts);
printf(" mpacket->size: %d\n", m_packet->size);
if (m_recording) {
int err = 0;
//err = avcodec_send_packet(m_output_video_codec_context, m_packet);
printf(" encoding error: %d\n", err);
}
}
}
// We're done with the packet (it's been unpacked to a frame), so deallocate & reset to defaults:
/*
if (m_frame == NULL)
return false;
if (m_frame->data[0] == NULL || m_frame->data[1] == NULL || m_frame->data[2] == NULL) {
printf("WARNING: null frame data");
continue;
}
*/
}
}
int MediaContainerMgr::decode_packet() {
// Supply raw packet data as input to a decoder
// https://ffmpeg.org/doxygen/trunk/group__lavc__decoding.html#ga58bc4bf1e0ac59e27362597e467efff3
int response = avcodec_send_packet(m_codec_context, m_packet);
if (response < 0) {
char buf[256];
av_strerror(response, buf, 256);
printf("Error while receiving a frame from the decoder: %s\n", buf);
return response;
}
// Return decoded output data (into a frame) from a decoder
// https://ffmpeg.org/doxygen/trunk/group__lavc__decoding.html#ga11e6542c4e66d3028668788a1a74217c
response = avcodec_receive_frame(m_codec_context, m_frame);
if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
return response;
} else if (response < 0) {
char buf[256];
av_strerror(response, buf, 256);
printf("Error while receiving a frame from the decoder: %s\n", buf);
return response;
} else {
printf(
"Frame %d (type=%c, size=%d bytes) pts %lld key_frame %d [DTS %d]\n",
m_codec_context->frame_number,
av_get_picture_type_char(m_frame->pict_type),
m_frame->pkt_size,
m_frame->pts,
m_frame->key_frame,
m_frame->coded_picture_number
);
}
return 0;
}
bool MediaContainerMgr::init_video_output(const std::string& video_file_name, unsigned int width, unsigned int height) {
if (m_recording)
return true;
m_recording = true;
advance_to(0L); // I've deleted the implmentation. Just seeks to beginning of vid. Works fine.
if (!(m_output_format = av_guess_format(nullptr, video_file_name.c_str(), nullptr))) {
printf("Cannot guess output format.\n");
return false;
}
int err = avformat_alloc_output_context2(&m_output_format_context, m_output_format, nullptr, video_file_name.c_str());
if (err < 0) {
printf("Failed to allocate output context.\n");
return false;
}
//TODO(P0): Break out the video and audio inits into their own methods.
m_output_video_codec = avcodec_find_encoder(m_output_format->video_codec);
if (!m_output_video_codec) {
printf("Failed to create video codec.\n");
return false;
}
m_output_video_stream = avformat_new_stream(m_output_format_context, m_output_video_codec);
if (!m_output_video_stream) {
printf("Failed to find video format.\n");
return false;
}
m_output_video_codec_context = avcodec_alloc_context3(m_output_video_codec);
if (!m_output_video_codec_context) {
printf("Failed to create video codec context.\n");
return(false);
}
m_output_video_stream->codecpar->codec_id = m_output_format->video_codec;
m_output_video_stream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
m_output_video_stream->codecpar->width = width;
m_output_video_stream->codecpar->height = height;
m_output_video_stream->codecpar->format = AV_PIX_FMT_YUV420P;
// Use the same bit rate as the input stream.
m_output_video_stream->codecpar->bit_rate = m_format_context->streams[m_video_stream_index]->codecpar->bit_rate;
m_output_video_stream->avg_frame_rate = m_format_context->streams[m_video_stream_index]->avg_frame_rate;
avcodec_parameters_to_context(m_output_video_codec_context, m_output_video_stream->codecpar);
m_output_video_codec_context->time_base = m_format_context->streams[m_video_stream_index]->time_base;
//TODO(P1): Set these to match the input stream?
m_output_video_codec_context->max_b_frames = 2;
m_output_video_codec_context->gop_size = 12;
m_output_video_codec_context->framerate = m_format_context->streams[m_video_stream_index]->r_frame_rate;
//m_output_codec_context->refcounted_frames = 0;
if (m_output_video_stream->codecpar->codec_id == AV_CODEC_ID_H264) {
av_opt_set(m_output_video_codec_context, "preset", "ultrafast", 0);
} else if (m_output_video_stream->codecpar->codec_id == AV_CODEC_ID_H265) {
av_opt_set(m_output_video_codec_context, "preset", "ultrafast", 0);
} else {
av_opt_set_int(m_output_video_codec_context, "lossless", 1, 0);
}
avcodec_parameters_from_context(m_output_video_stream->codecpar, m_output_video_codec_context);
m_output_audio_codec = avcodec_find_encoder(m_output_format->audio_codec);
if (!m_output_audio_codec) {
printf("Failed to create audio codec.\n");
return false;
}
I've commented out all of the audio stream init beyond this next line, because this is where
the trouble begins. Creating this output stream causes the null reference I mentioned. If I
uncomment everything below here, I still get the null deref. If I comment out this line, the
deref exception vanishes. (IOW, I commented out more and more code until I found that this
was the trigger that caused the problem.)
I assume that there's something I'm doing wrong in the rest of the commented out code, that,
when fixed, will fix the nullptr and give me a working audio stream.
m_output_audio_stream = avformat_new_stream(m_output_format_context, m_output_audio_codec);
if (!m_output_audio_stream) {
printf("Failed to find audio format.\n");
return false;
}
/*
m_output_audio_codec_context = avcodec_alloc_context3(m_output_audio_codec);
if (!m_output_audio_codec_context) {
printf("Failed to create audio codec context.\n");
return(false);
}
m_output_audio_stream->codecpar->codec_id = m_output_format->audio_codec;
m_output_audio_stream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
m_output_audio_stream->codecpar->format = m_format_context->streams[m_audio_stream_index]->codecpar->format;
m_output_audio_stream->codecpar->bit_rate = m_format_context->streams[m_audio_stream_index]->codecpar->bit_rate;
m_output_audio_stream->avg_frame_rate = m_format_context->streams[m_audio_stream_index]->avg_frame_rate;
avcodec_parameters_to_context(m_output_audio_codec_context, m_output_audio_stream->codecpar);
m_output_audio_codec_context->time_base = m_format_context->streams[m_audio_stream_index]->time_base;
*/
//TODO(P2): Free assets that have been allocated.
err = avcodec_open2(m_output_video_codec_context, m_output_video_codec, nullptr);
if (err < 0) {
printf("Failed to open codec.\n");
return false;
}
if (!(m_output_format->flags & AVFMT_NOFILE)) {
err = avio_open(&m_output_format_context->pb, video_file_name.c_str(), AVIO_FLAG_WRITE);
if (err < 0) {
printf("Failed to open output file.");
return false;
}
}
err = avformat_write_header(m_output_format_context, NULL);
if (err < 0) {
printf("Failed to write header.\n");
return false;
}
av_dump_format(m_output_format_context, 0, video_file_name.c_str(), 1);
return true;
}
//TODO(P2): make this a member. (Thanks to https://emvlo.wordpress.com/2016/03/10/sws_scale/)
void PrepareFlipFrameJ420(AVFrame* pFrame) {
for (int i = 0; i < 4; i++) {
if (i)
pFrame->data[i] += pFrame->linesize[i] * ((pFrame->height >> 1) - 1);
else
pFrame->data[i] += pFrame->linesize[i] * (pFrame->height - 1);
pFrame->linesize[i] = -pFrame->linesize[i];
}
}
This is where we take an altered frame and write it to the output container. This works fine
as long as we haven't set up an audio stream in the output container.
bool MediaContainerMgr::output_video_frame(uint8_t* buf) {
int err;
if (!m_output_video_frame) {
m_output_video_frame = av_frame_alloc();
m_output_video_frame->format = AV_PIX_FMT_YUV420P;
m_output_video_frame->width = m_output_video_codec_context->width;
m_output_video_frame->height = m_output_video_codec_context->height;
err = av_frame_get_buffer(m_output_video_frame, 32);
if (err < 0) {
printf("Failed to allocate output frame.\n");
return false;
}
}
if (!m_output_scale_context) {
m_output_scale_context = sws_getContext(m_output_video_codec_context->width, m_output_video_codec_context->height,
AV_PIX_FMT_RGB24,
m_output_video_codec_context->width, m_output_video_codec_context->height,
AV_PIX_FMT_YUV420P, SWS_BICUBIC, nullptr, nullptr, nullptr);
}
int inLinesize[1] = { 3 * m_output_video_codec_context->width };
sws_scale(m_output_scale_context, (const uint8_t* const*)&buf, inLinesize, 0, m_output_video_codec_context->height,
m_output_video_frame->data, m_output_video_frame->linesize);
PrepareFlipFrameJ420(m_output_video_frame);
//TODO(P0): Switch m_frame to be m_input_video_frame so I don't end up using the presentation timestamp from
// an audio frame if I threadify the frame reading.
m_output_video_frame->pts = m_frame->pts;
printf("Output PTS: %d, time_base: %d/%d\n", m_output_video_frame->pts,
m_output_video_codec_context->time_base.num, m_output_video_codec_context->time_base.den);
err = avcodec_send_frame(m_output_video_codec_context, m_output_video_frame);
if (err < 0) {
printf(" ERROR sending new video frame output: ");
switch (err) {
case AVERROR(EAGAIN):
printf("AVERROR(EAGAIN): %d\n", err);
break;
case AVERROR_EOF:
printf("AVERROR_EOF: %d\n", err);
break;
case AVERROR(EINVAL):
printf("AVERROR(EINVAL): %d\n", err);
break;
case AVERROR(ENOMEM):
printf("AVERROR(ENOMEM): %d\n", err);
break;
}
return false;
}
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = nullptr;
pkt.size = 0;
pkt.flags |= AV_PKT_FLAG_KEY;
int ret = 0;
if ((ret = avcodec_receive_packet(m_output_video_codec_context, &pkt)) == 0) {
static int counter = 0;
printf("pkt.key: 0x%08x, pkt.size: %d, counter:\n", pkt.flags & AV_PKT_FLAG_KEY, pkt.size, counter++);
uint8_t* size = ((uint8_t*)pkt.data);
printf("sizes: %d %d %d %d %d %d %d %d %d\n", size[0], size[1], size[2], size[2], size[3], size[4], size[5], size[6], size[7]);
av_interleaved_write_frame(m_output_format_context, &pkt);
}
printf("push: %d\n", ret);
av_packet_unref(&pkt);
return true;
}
bool MediaContainerMgr::finalize_output() {
if (!m_recording)
return true;
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = nullptr;
pkt.size = 0;
for (;;) {
avcodec_send_frame(m_output_video_codec_context, nullptr);
if (avcodec_receive_packet(m_output_video_codec_context, &pkt) == 0) {
av_interleaved_write_frame(m_output_format_context, &pkt);
printf("final push:\n");
} else {
break;
}
}
av_packet_unref(&pkt);
av_write_trailer(m_output_format_context);
if (!(m_output_format->flags & AVFMT_NOFILE)) {
int err = avio_close(m_output_format_context->pb);
if (err < 0) {
printf("Failed to close file. err: %d\n", err);
return false;
}
}
return true;
}
EDIT
The call stack on the crash (which I should have included in the original question):
avformat-58.dll!compute_muxer_pkt_fields(AVFormatContext * s, AVStream * st, AVPacket * pkt) Line 630 C
avformat-58.dll!write_packet_common(AVFormatContext * s, AVStream * st, AVPacket * pkt, int interleaved) Line 1122 C
avformat-58.dll!write_packets_common(AVFormatContext * s, AVPacket * pkt, int interleaved) Line 1186 C
avformat-58.dll!av_interleaved_write_frame(AVFormatContext * s, AVPacket * pkt) Line 1241 C
CamBot.exe!MediaContainerMgr::output_video_frame(unsigned char * buf) Line 553 C++
CamBot.exe!main() Line 240 C++
If I move the call to avformat_write_header so it's immediately before the audio stream initialization, I still get a crash, but in a different place. The crash happens on line 6459 of movenc.c, where we have:
/* Non-seekable output is ok if using fragmentation. If ism_lookahead
* is enabled, we don't support non-seekable output at all. */
if (!(s->pb->seekable & AVIO_SEEKABLE_NORMAL) && // CRASH IS HERE
(!(mov->flags & FF_MOV_FLAG_FRAGMENT) || mov->ism_lookahead)) {
av_log(s, AV_LOG_ERROR, "muxer does not support non seekable output\n");
return AVERROR(EINVAL);
}
The exception is a nullptr exception, where s->pb is NULL. The call stack is:
avformat-58.dll!mov_init(AVFormatContext * s) Line 6459 C
avformat-58.dll!init_muxer(AVFormatContext * s, AVDictionary * * options) Line 407 C
[Inline Frame] avformat-58.dll!avformat_init_output(AVFormatContext *) Line 489 C
avformat-58.dll!avformat_write_header(AVFormatContext * s, AVDictionary * * options) Line 512 C
CamBot.exe!MediaContainerMgr::init_video_output(const std::string & video_file_name, unsigned int width, unsigned int height) Line 424 C++
CamBot.exe!main() Line 183 C++
Please note that you should always try to provide a self-contained minimal working example to make it easier for others to help. With the actual code, the matching FFmpeg version, and an input video that triggers the segmentation fault (to be sure), the issue would be a matter of analyzing the control flow to identify why st->internal->priv_pts was not allocated. Without the full scenario, I have to report to making assumptions that may or may not correspond to your actual code.
Based on your description, I attempted to reproduce the issue by cloning https://github.com/FFmpeg/FFmpeg.git and creating a new branch from commit b52e0d95 (November 4, 2020) to approximate your FFmpeg version.
I recreated your scenario using the provided code snippets by
including the avformat_new_stream() call for the audio stream
keeping the remaining audio initialization commented out
including the original avformat_write_header() call site (unchanged order)
With that scenario, the video write with MP4 video/audio input fails in avformat_write_header():
[mp4 # 0x2b39f40] sample rate not set 0
The call stack of the error location:
#0 0x00007ffff75253d7 in raise () from /lib64/libc.so.6
#1 0x00007ffff7526ac8 in abort () from /lib64/libc.so.6
#2 0x000000000094feca in init_muxer (s=0x2b39f40, options=0x0) at libavformat/mux.c:309
#3 0x00000000009508f4 in avformat_init_output (s=0x2b39f40, options=0x0) at libavformat/mux.c:490
#4 0x0000000000950a10 in avformat_write_header (s=0x2b39f40, options=0x0) at libavformat/mux.c:514
[...]
In init_muxer(), the sample rate in the stream parameters is checked unconditionally:
case AVMEDIA_TYPE_AUDIO:
if (par->sample_rate <= 0) {
av_log(s, AV_LOG_ERROR, "sample rate not set %d\n", par->sample_rate); abort();
ret = AVERROR(EINVAL);
goto fail;
}
That condition has been in effect since 2014-06-18 at the very least (didn't go back any further) and still exists. With a version from November 2020, the check must be active and the parameter must be set accordingly.
If I uncomment the remaining audio initialization, the situation remains unchanged (as expected). So, satisfy the condition, I added the missing parameter as follows:
m_output_audio_stream->codecpar->sample_rate =
m_format_context->streams[m_audio_stream_index]->codecpar->sample_rate;
With that, the check succeeds, avformat_write_header() succeeds, and the actual video write succeeds.
As you indicated in your question, the segmentation fault is caused by st->internal->priv_pts being NULL at this location:
#0 0x00000000009516db in compute_muxer_pkt_fields (s=0x2b39f40, st=0x2b3a580, pkt=0x7fffffffe2d0) at libavformat/mux.c:632
#1 0x0000000000953128 in write_packet_common (s=0x2b39f40, st=0x2b3a580, pkt=0x7fffffffe2d0, interleaved=1) at libavformat/mux.c:1125
#2 0x0000000000953473 in write_packets_common (s=0x2b39f40, pkt=0x7fffffffe2d0, interleaved=1) at libavformat/mux.c:1188
#3 0x0000000000953634 in av_interleaved_write_frame (s=0x2b39f40, pkt=0x7fffffffe2d0) at libavformat/mux.c:1243
[...]
In the FFmpeg code base, the allocation of priv_pts is handled by init_pts() for all streams referenced by the context. init_pts() has two call sites:
libavformat/mux.c:496:
if (s->oformat->init && ret) {
if ((ret = init_pts(s)) < 0)
return ret;
return AVSTREAM_INIT_IN_INIT_OUTPUT;
}
libavformat/mux.c:530:
if (!s->internal->streams_initialized) {
if ((ret = init_pts(s)) < 0)
goto fail;
}
In both cases, the calls are triggered by avformat_write_header() (indirectly via avformat_init_output() for the first, directly for the second). According to control flow analysis, there's no success case that would leave priv_pts unallocated.
Considering a high probability that our versions of FFmpeg are compatible in terms of behavior, I have to assume that 1) the sample rate must be provided for audio streams and 1) priv_pts is always allocated by avformat_write_header() in the absence of errors. Therefore, two possible root causes come to mind:
Your stream is not an audio stream (unlikely; the type is based on the codec, which in turn is based on the output file extension - assuming mp4)
You do not call avformat_write_header() (unlikely) or do not handle the error in the caller of your C++ member function (the return value of avformat_write_header() is checked but I do not have code corresponding to the caller of the C++ member function; your actual code might differ significantly from the code provided, so it's possible and the only plausible conclusion that can be drawn from available data)
The solution: Ensure that processing does not continue if avformat_write_header() fails. By adding the audio stream, avformat_write_header() starts to fail unless you set the stream sample rate. If the error is ignored, av_interleaved_write_frame() triggers a segmentation fault by accessing the unallocated st->internal->priv_pts.
As mentioned initially, scenario is incomplete. If you do call avformat_write_header() and stop processing in case of an error (meaning you do not call av_interleaved_write_frame()), more information is needed. As it stands now, that is unlikely. For further analysis, the executable output (stdout, stderr) is required to see your traces and FFmpeg log messages. If that does not reveal new information, a self-contained minimal working example and the video input are needed to get all the full picture.
Here is my audio init code. My app responds when queue buffers are ready, but all data in buffer is zero. Checking sound in system preferences shows that USB Audio CODEC in sound input dialog is active. AudioInit() is called right after app launches.
{
#pragma mark user data struct
typedef struct MyRecorder
{
AudioFileID recordFile;
SInt64 recordPacket;
Float32 *pSampledData;
MorseDecode *pMorseDecoder;
} MyRecorder;
#pragma mark utility functions
void CheckError(OSStatus error, const char *operation)
{
if(error == noErr) return;
char errorString[20];
// see if it appears to be a 4 char code
*(UInt32*)(errorString + 1) = CFSwapInt32HostToBig(error);
if (isprint(errorString[1]) && isprint(errorString[2]) &&
isprint(errorString[3]) && isprint(errorString[4]))
{
errorString[0] = errorString[5] = '\'';
errorString[6] = '\0';
}
else
{
sprintf(errorString, "%d", (int)error);
}
fprintf(stderr, "Error: %s (%s)\n", operation, errorString);
}
OSStatus MyGetDefaultInputDeviceSampleRate(Float64 *outSampleRate)
{
OSStatus error;
AudioDeviceID deviceID = 0;
AudioObjectPropertyAddress propertyAddress;
UInt32 propertySize;
propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = 0;
propertySize = sizeof(AudioDeviceID);
error = AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddress,
0,
NULL,
&propertySize,
&deviceID);
if(error)
return error;
propertyAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = 0;
propertySize = sizeof(Float64);
error = AudioObjectGetPropertyData(deviceID,
&propertyAddress,
0,
NULL,
&propertySize,
outSampleRate);
return error;
}
static int MyComputeRecordBufferSize(const AudioStreamBasicDescription *format,
AudioQueueRef queue,
float seconds)
{
int packets, frames, bytes;
frames = (int)ceil(seconds * format->mSampleRate);
if(format->mBytesPerFrame > 0)
{
bytes = frames * format->mBytesPerFrame;
}
else
{
UInt32 maxPacketSize;
if(format->mBytesPerPacket > 0)
{
// constant packet size
maxPacketSize = format->mBytesPerPacket;
}
else
{
// get the largest single packet size possible
UInt32 propertySize = sizeof(maxPacketSize);
CheckError(AudioQueueGetProperty(queue,
kAudioConverterPropertyMaximumOutputPacketSize,
&maxPacketSize,
&propertySize),
"Couldn't get queues max output packet size");
}
if(format->mFramesPerPacket > 0)
packets = frames / format->mFramesPerPacket;
else
// worst case scenario: 1 frame in a packet
packets = frames;
// sanity check
if(packets == 0)
packets = 1;
bytes = packets * maxPacketSize;
}
return bytes;
}
extern void bridgeToMainThread(MorseDecode *pDecode);
static int callBacks = 0;
// ---------------------------------------------
static void MyAQInputCallback(void *inUserData,
AudioQueueRef inQueue,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription *inPacketDesc)
{
MyRecorder *recorder = (MyRecorder*)inUserData;
Float32 *pAudioData = (Float32*)(inBuffer->mAudioData);
recorder->pMorseDecoder->pBuffer = pAudioData;
recorder->pMorseDecoder->bufferSize = inNumPackets;
bridgeToMainThread(recorder->pMorseDecoder);
CheckError(AudioQueueEnqueueBuffer(inQueue,
inBuffer,
0,
NULL),
"AudioQueueEnqueueBuffer failed");
printf("packets = %ld, bytes = %ld\n",(long)inNumPackets,(long)inBuffer->mAudioDataByteSize);
callBacks++;
//printf("\ncallBacks = %d\n",callBacks);
//if(callBacks == 0)
//audioStop();
}
static AudioQueueRef queue = {0};
static MyRecorder recorder = {0};
static AudioStreamBasicDescription recordFormat;
void audioInit()
{
// set up format
memset(&recordFormat,0,sizeof(recordFormat));
recordFormat.mFormatID = kAudioFormatLinearPCM;
recordFormat.mChannelsPerFrame = 2;
recordFormat.mBitsPerChannel = 32;
recordFormat.mBytesPerPacket = recordFormat.mBytesPerFrame = recordFormat.mChannelsPerFrame * sizeof(Float32);
recordFormat.mFramesPerPacket = 1;
//recordFormat.mFormatFlags = kAudioFormatFlagsCanonical;
recordFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
MyGetDefaultInputDeviceSampleRate(&recordFormat.mSampleRate);
UInt32 propSize = sizeof(recordFormat);
CheckError(AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
0,
NULL,
&propSize,
&recordFormat),
"AudioFormatProperty failed");
recorder.pMorseDecoder = MorseDecode::pInstance();
recorder.pMorseDecoder->m_sampleRate = recordFormat.mSampleRate;
// recorder.pMorseDecoder->setCircularBuffer();
//set up queue
CheckError(AudioQueueNewInput(&recordFormat,
MyAQInputCallback,
&recorder,
NULL,
kCFRunLoopCommonModes,
0,
&queue),
"AudioQueueNewInput failed");
UInt32 size = sizeof(recordFormat);
CheckError(AudioQueueGetProperty(queue,
kAudioConverterCurrentOutputStreamDescription,
&recordFormat,
&size), "Couldn't get queue's format");
// set up buffers and enqueue
const int kNumberRecordBuffers = 3;
int bufferByteSize = MyComputeRecordBufferSize(&recordFormat, queue, AUDIO_BUFFER_DURATION);
for(int bufferIndex = 0; bufferIndex < kNumberRecordBuffers; bufferIndex++)
{
AudioQueueBufferRef buffer;
CheckError(AudioQueueAllocateBuffer(queue,
bufferByteSize,
&buffer),
"AudioQueueAllocateBuffer failed");
CheckError(AudioQueueEnqueueBuffer(queue,
buffer,
0,
NULL),
"AudioQueueEnqueueBuffer failed");
}
}
void audioRun()
{
CheckError(AudioQueueStart(queue, NULL), "AudioQueueStart failed");
}
void audioStop()
{
CheckError(AudioQueuePause(queue), "AudioQueuePause failed");
}
}
This sounds like the new macOS 'microphone privacy' setting, which, if set to 'no access' for your app, will cause precisely this behaviour. So:
Open the System Preferences pane.
Click on 'Security and Privacy'.
Select the Privacy tab.
Click on 'Microphone' in the left-hand pane.
Locate your app in the right-hand pane and tick the checkbox next to it.
Then restart your app and test it.
Tedious, no?
Edit: As stated in the comments, you can't directly request microphone access, but you can detect whether it has been granted to your app or not by calling [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio].
I'm trying to write a C++ program (altough python would've been fine as well in case someone knows a better (IAX/SIP) alternative) which connects to an Asterisk server.
After connecting, it should listen for audio and process that. It should also send audio back. I'm using https://sourceforge.net/projects/iaxclient/ for that (note that there are several versions (betas, regular releases, svn version) which all behave differently).
Now if I understood the code of the library correct, then it can call a callback function with an event. One of those events is IAXC_EVENT_AUDIO. In the structure of that IAXC_EVENT_AUDIO there's a direction; incoming outgoing. And that's where I'm lost: with some versions of iaxclient I only receive the IAXC_SOURCE_REMOTE messages, with some both. And if I switch to test-mode (which should only disable the audio-device) I often receive nothing at all. When I receive both IAXC_SOURCE_LOCAL and IAXC_SOURCE_REMOTE, I tried to set the buffers of those events to random data but that doesn't reach the other end at all (I set it to RAW mode).
As anyone any suggestions how to resolve this?
My test-code is:
#include <iaxclient.h>
#include <unistd.h>
int iaxc_event_callback(iaxc_event e)
{
if (e.type == IAXC_EVENT_TEXT) {
printf("text\n");
}
else if (e.type == IAXC_EVENT_LEVELS) {
printf("level\n");
}
else if (e.type == IAXC_EVENT_STATE) {
struct iaxc_ev_call_state *st = iaxc_get_event_state(&e);
printf("\tcallno %d state %d format %d remote %s(%s)\n", st->callNo, st->state, st->format,st->remote, st->remote_name);
iaxc_key_radio(st->callNo);
}
else if (e.type == IAXC_EVENT_NETSTAT) {
printf("\tcallno %d rtt %d\n", e.ev.netstats.callNo, e.ev.netstats.rtt);
}
else if (e.type == IAXC_EVENT_AUDIO) {
printf("\t AUDIO!!!! %d %u %d\n", e.ev.audio.source, e.ev.audio.ts, e.ev.audio.size);
for(int i=0; i<e.ev.audio.size; i++)
printf("%02x ", e.ev.audio.data[i]);
printf("\n");
}
else {
printf("type: %d\n", e.type);
}
return 1;
}
int main(int argc, char *argv[])
{
iaxc_set_test_mode(1);
printf("init %d\n", iaxc_initialize(1));
iaxc_set_formats(IAXC_FORMAT_SPEEX, IAXC_FORMAT_SPEEX);
iaxc_set_event_callback(iaxc_event_callback);
printf("get audio pref %d\n", iaxc_get_audio_prefs());
//printf("set audio pref %d\n", iaxc_set_audio_prefs(IAXC_AUDIO_PREF_RECV_REMOTE_ENCODED));
printf("set audio pref %d\n", iaxc_set_audio_prefs(IAXC_AUDIO_PREF_RECV_REMOTE_RAW | IAXC_AUDIO_PREF_RECV_LOCAL_RAW));
printf("get audio pref %d\n", iaxc_get_audio_prefs());
printf("start thread %d\n", iaxc_start_processing_thread());
int id = -1;
printf("register %d\n", id = iaxc_register("6003", "1923", "192.168.64.1"));
int callNo = -1;
printf("call %d\n", callNo = iaxc_call("6003:1923#192.168.64.1/6001"));
printf("unquelch: %d\n", iaxc_unquelch(callNo));
pause();
printf("finish\n");
printf("%d\n", iaxc_unregister(id));
printf("%d\n", iaxc_stop_processing_thread());
iaxc_shutdown();
return 0;
}
Please have a look in iaxclient_lib.c too see how the logic works. To hook or replace input/output you can change function iaxci_do_audio_callback at memcpy(e.ev.audio.data, data, size); where the buffer is set. Also have a look at service_audio to understand how you can replace the buffer/stream sent to the remote location (e.g. want_send_audio and want_local_audio). You can also create virtual input/output devices in portaudio that iaxclient use for process audio using buffers instead.
For a more concrete example please look at the main method in the simplecall source to get a good start. However, the source code is to long for copy and paste, sorry about that.
I have created a Win32 Console Application project in Visual Studio 2012 C++.
How can I substitute the Console window by a more appealing GUI like Windows Forms?
int32_t main(int32_t argc, char* argv[])
{
const char *date = "20150428_1\\";
int mode=0;
_CallServerPtr pCallServer;
uint32_t start_address_comp=0;
uint32_t start_address_module=0;
const char* xmlFile_tx_dbb="tx_dbb.xml";;
char str[100] = "\0";
char localeStr[64];
memset(localeStr, 0, sizeof localeStr);
const char *l_path = "..\\XERCES\\Configs\\";
std::string buf = "";
double Fsym_Hz=(1/1.15)*1e9;
int selection=0;
int user_selection=0;
try
{
if (strlen(localeStr))
{
XMLPlatformUtils::Initialize(localeStr);
}
else
{
XMLPlatformUtils::Initialize();
}
}
catch (const XMLException& toCatch)
{
XERCES_STD_QUALIFIER cerr << "Error during initialization! :\n"
<< StrX(toCatch.getMessage()) << XERCES_STD_QUALIFIER endl;
}
static const XMLCh gLS[] = { chLatin_L, chLatin_S, chNull };
DOMImplementation *impl = DOMImplementationRegistry::getDOMImplementation(gLS);
DOMLSParser *parser = ((DOMImplementationLS*)impl)->createLSParser(DOMImplementationLS::MODE_SYNCHRONOUS, 0);
DOMConfiguration *config = parser->getDomConfig();
DOMLSSerializer *theSerializer = ((DOMImplementationLS*)impl)->createLSSerializer();
DOMLSOutput *theOutputDesc = ((DOMImplementationLS*)impl)->createLSOutput();
config->setParameter(XMLUni::fgDOMDatatypeNormalization, true);
DOMCountErrorHandler errorHandler;
config->setParameter(XMLUni::fgDOMErrorHandler, &errorHandler);
XERCES_STD_QUALIFIER ifstream fin;
//reset error count first
errorHandler.resetErrors();*/
// reset document pool
parser->resetDocumentPool();
char* pszHostname = NULL;
pSaIn = 0;
pSaOut = 0;
// Initialize the COM Library
CoInitialize(NULL);
if (!pszHostname)
{
// Create the CallServer server object on the local computer
pCallServer.CreateInstance(CLSID_CallServer);
}
if (pCallServer == NULL)
throw "Failed to create the CallableVEE CallServer object";
// Load the VEE User Function library
char strpath[256];
strcpy (strpath,reposity_path);
strcat (strpath,l_path_vee);
_bstr_t bstrLibPath(strpath);
LibraryPtr pLib = pCallServer->GetLibraries()->Load(bstrLibPath);
// Print out the names of the UserFunctions in this library.
UserFunctionsPtr pUserFuncColl = pLib->GetUserFunctions();
VARIANT_BOOL bDebug = VARIANT_FALSE;
pCallServer->PutDebug(bDebug);
// Variables added by ivi
float *freq =(float *)_aligned_malloc(6,16); // Read frequency vector
// Previous variables
int32_t devIdx;
int32_t modeClock;
int32_t ifType;
const char *devType;
char fpga_device_type[32];
int32_t rc;
int32_t ref_clk=0;
uint32_t carrier=0;
uint32_t odelay_dac0 = 0;
uint32_t odelay_dac1 = 0;
// Parse the application arguments
if(argc!=5) {
printf("Usage: FMCxxxApp.exe {interface type} {device type} {device index} {clock mode} \n\n");
printf(" {interface type} can be either 0 (PCI) or 1 (Ethernet). At CEIT, we use 1 (Ethernet).\n");
printf(" {device type} is a string defining the target hardware (VP680, ML605, ...). At CEIT, we use VC707.\n");
printf(" {device index} is a PCI index or an Ethernet interface index. This value depends on the PC.\n");
printf(" {clock mode} can be either 0 (Int. Clock) or 1 (Ext. Clock)\n");
printf("\n");
printf("\n");
printf("Example: Fmc230APP.exe 1 VC707 0 0\n");
printf("\n");
printf("\n");
printf(" List of NDIS interfaces found in the system {device index}:\n");
printf(" -----------------------------------------------------------\n");
if(sipif_getdeviceenumeration(API_ENUM_DISPLAY)!=SIPIF_ERR_OK) {
printf("Could not obtain NDIS(Ethernet) device enumeration...\n Check if the 4dspnet driver installed or if the service started?\n");
printf("You can discard this error if you do not have any Ethernet based product in use.");
}
if( EXIT_IF_ERRORS)
{
sipif_free();
system("pause");
return -1;
}
...
}
You mean to have the same code in windows forms. That won't work. The printf and other commands work only in a console application. You must to create a windows form application and rewrite the code for it. You must rewrite all commands that don't work in a windows form application. There probably exists a conversion application, but for this short code I think it's better to rewrite it.
ListenSocket.h
// class does not contain WSASTARTUP () and WSACLEANUP ()
#ifndef LISTENTHREAD_H
#define LISTENTHREAD_H
#include "includes.h"
#include "LOGMSGs.h"
// 1, CListenSocket: class is used to create the listener thread local variable.
// This class can be reused. When you call Close () is closed, re-calling Open () the new listening port. But the system did not use the feature.
class CListenSocket
{
public:
// main method:
// BIND each object only to a port.
CListenSocket(u_short nPort, int nSndSize = 0);
// to release SOCKET
~CListenSocket(){};
// Create server listening SOCKET, specific options see the code. Fails to return false.
bool Open(); // call can be repeated
// error return INVALID_SOCKET
SOCKET Accept(u_long & nClientIP);
// repeated calls. Usually not, can be used to take the initiative to close the SOCKET.
// close the re-call after Open () re-use the object.
void Close(); // call can be repeated
bool IsOpen() { return m_bState; }
bool Rebuild();
public:
SOCKET Socket() { return m_sockListen; }
protected:
// main member variables:
const u_short m_nPort;
const int m_nSndBuf;
SOCKET m_sockListen;
// network status is normal sign.
// When the value is false that the object is not available. May not have Open (), may also be a network error.
bool m_bState;
time_t m_tCloseTime; // SOCKET last closed the time delay for the automatic re-SOCKET
};
#endif // LISTENTHREAD_H
ListenSocket.cpp
#include "ListenSocket.h"
long s_nSocketCount = 0;
int REBUILDLISTENDELAYSEC;
CListenSocket::CListenSocket(u_short nPort, int nSndBuf /*= 0*/) // 0: Default
: m_nPort(nPort), m_nSndBuf(nSndBuf)
{
m_sockListen = INVALID_SOCKET;
m_bState = false;
// m_nPort = nPort;
m_tCloseTime = 0;
}
// Error returned INVALID_SOCKET
SOCKET CListenSocket::Accept(u_long & nClientIP)
{
/*
// Reconstruction SOCKET
if(!m_bState)
{
if(clock() < m_tCloseTime + REBUILDLISTENDELAYSEC*CLOCKS_PER_SEC)
return INVALID_SOCKET;
else
{
LOGMSG("Anti-crash system start listening SOCKET [%d] re under construction...", m_nPort);
if(Open())
{
LOGMSG("... listen SOCKET reconstruction success.");
PrintText("Listen SOCKET [%d] failed to rebuild SOCKET success. Server continues to run in the ...", m_nPort);
}
else
{
Error("... listen SOCKET reconstruction has failed. Server will not accept new connections");
PrintText("Listen SOCKET [%d] error, [%d] seconds after the re-SOCKET. Server continues to run in the ...", m_nPort, REBUILDLISTENDELAYSEC); // nDelaySec);
}
m_tCloseTime = clock();
}
}
//*/
if(!m_bState)
{
Error("ACCEPT inner exception a1");
return INVALID_SOCKET;
}
// ACCEPT
struct sockaddr_in addr;
memset(&addr, 0, sizeof(addr));
int len = sizeof(addr);
SOCKET newsock = accept(m_sockListen, (sockaddr*)&addr, (int*)&len); // receive to the other side of the map, you can use
#ifdef PROFILE_X
// Analysis Accept speed (cycle speed)
const int nTimes2 = ACCEPTPROFILESEC; // Statistics once every 30 seconds the speed ACCEPT
static clock_t tNextTime2 = clock() + nTimes2 * CLOCKS_PER_SEC; //? Only one monitor thread, no sharing violation
static long nCount2 = 0; //? Only one monitor thread, no sharing violation
if(clock() >= tNextTime2)
{
PrintText("Each [%d] seconds to execute a [%d] times Accept ()", nTimes2, InterlockedExchange(&nCount2, 0));
tNextTime2 = clock() + nTimes2 * CLOCKS_PER_SEC;
}
else
{
InterlockedIncrement(&nCount2);
}
#endif // PROFILE
if(newsock == INVALID_SOCKET)
{
// Network Error
int err = WSAGetLastError();
if(err != WSAEWOULDBLOCK)
{
PrintText("Listen SOCKET %d failed, %s seconds after the re-SOCKET.", m_nPort, REBUILDLISTENDELAYSEC);
Error("Listen SOCKET [%d] failed [%d], [%s] seconds after the re-SOCKET.", m_nPort, err, REBUILDLISTENDELAYSEC);
Close();
}
else
Error("ACCEPT inner exception a2");
return INVALID_SOCKET;
}
else
{
nClientIP = addr.sin_addr.S_un.S_addr;
InterlockedIncrement(&s_nSocketCount);
}
// Check whether the SOCKET closed
fd_set readmask;
FD_ZERO(&readmask);
FD_SET(newsock, &readmask);
struct timeval timeout = {0, 0};
/*
char nTemp;
if(select(FD_SETSIZE, &readmask, (fd_set *) 0, (fd_set *) 0, &timeout)
&& recv(newsock, &nTemp, 1, MSG_PEEK) == 0)
{
#ifdef ALPHA_X
LOGMSG("ACCEPT a new SOCKET is invalid .");
#endif
closesocket(newsock);
InterlockedDecrement(&s_nSocketCount);
return INVALID_SOCKET;
}
//else*/
//*
fd_set exceptmask;
FD_ZERO(&exceptmask);
FD_SET(newsock, &exceptmask);
int ret = select(FD_SETSIZE, &readmask, (fd_set *) 0, (fd_set *) &exceptmask, &timeout);
if(ret < 0)
{
Error("ACCEPT a new SOCKET is invalid . can't read"); // Not trigger
closesocket(newsock);
InterlockedDecrement(&s_nSocketCount);
return INVALID_SOCKET;
}
else if(ret > 0)
{
if(FD_ISSET(newsock, &exceptmask))
{
LOGMSG("ACCEPT a new SOCKET is invalid.except"); // Not trigger
closesocket(newsock);
InterlockedDecrement(&s_nSocketCount);
return INVALID_SOCKET;
}
else if(FD_ISSET(newsock, &readmask))
{
char nTemp;
if(recv(newsock, &nTemp, 1, MSG_PEEK) == 0)
{
#ifdef ALPHA_X
LOGMSG("ACCEPT a new SOCKET is invalid. recv==0"); // Not trigger
#endif
closesocket(newsock);
InterlockedDecrement(&s_nSocketCount);
return INVALID_SOCKET;
}
}
}
//*/
#ifdef PROFILE_X
// analysis Accept speed (received valid SOCKET)
const int nTimes = ACCEPTPROFILESEC; // Statistics once every 10 seconds the speed ACCEPT
static clock_t tNextTime = clock() + nTimes * CLOCKS_PER_SEC; //? Only one monitor thread, no sharing violation
static long nCount = 0; //? Only one monitor thread, no sharing violation
if(clock() >= tNextTime)
{
LOGPROFILE("Port [%d] for every [%d] seconds, the successful implementation of the [%d] times Accept()",
m_nPort, nTimes, InterlockedExchange(&nCount, 0));
tNextTime = clock() + nTimes * CLOCKS_PER_SEC;
}
else
{
InterlockedIncrement(&nCount);
}
#endif // PROFILE
return newsock;
}
Main.cpp
#include "includes.h"
#include "IniFile.h"
#include "LOGMSGs.h"
#include "ListenSocket.h"
CListenSocket Sock(9985);
int main()
{
Sock.Open();
if(!Sock.Open())
{
Sock.Rebuild();
}
if(Sock.IsOpen())
PrintText("okey");
Sock.Socket();
u_long ip;
Sock.Accept(ip);
}
but i always got this error : ACCEPT inner exception a2 while it should work anyclue why?
CListenSocket Sock(9985);
int main()
{
Sock.Open();
if(!Sock.Open())
/* I think you meant 'IsOpen()' */
{
Sock.Rebuild();
}
if(Sock.IsOpen())
PrintText("okey");
Sock.Socket();
u_long ip;
Sock.Accept(ip);
}
Incidentally, this code sure reads funny. It feels like a generic toolkit programmed without a specific goal in mind. Maybe I'm missing it, but I have to think you'd have better results if you just wrote the network code that actually needed, and then abstract out the common bits into some helper routines later. There's no point in trying too hard to make the be-all and end-all network helper library, but there is a huge point in making tools that collapse common cases.
Feel free to ignore that last paragraph if you know what you're doing :) but if you're just starting out, I'd like to suggest writing a few smaller clients and servers, and then try writing your abstraction layer.