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.
I am using the c libraries of ffmpeg to read frames from a video and create an output file that is supposed to be identical to the input.
However, somewhere during this process some quality gets lost and the result is "less sharp". My guess is that the problem is the encoding and that the frames are too compressed (also because the size of the file decreases quite significantly). Is there some parameter in the encoder that allows me to control the quality of the result? I found that AVCodecContext has a compression_level member, but changing it that does not seem to have any effect.
I post here part of my code in case it could help. I would say that something must be changed in the init function of OutputVideoBuilder when I set the codec. The AVCodecContext that is passed to the method is the same of InputVideoHandler.
Here are the two main classes that I created to wrap the ffmpeg functionalities:
// This class opens the video files and sets the decoder
class InputVideoHandler {
public:
InputVideoHandler(char* name);
~InputVideoHandler();
AVCodecContext* getCodecContext();
bool readFrame(AVFrame* frame, int* success);
private:
InputVideoHandler();
void init(char* name);
AVFormatContext* formatCtx;
AVCodec* codec;
AVCodecContext* codecCtx;
AVPacket packet;
int streamIndex;
};
void InputVideoHandler::init(char* name) {
streamIndex = -1;
int numStreams;
if (avformat_open_input(&formatCtx, name, NULL, NULL) != 0)
throw std::exception("Invalid input file name.");
if (avformat_find_stream_info(formatCtx, NULL)<0)
throw std::exception("Could not find stream information.");
numStreams = formatCtx->nb_streams;
if (numStreams < 0)
throw std::exception("No streams in input video file.");
for (int i = 0; i < numStreams; i++) {
if (formatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
streamIndex = i;
break;
}
}
if (streamIndex < 0)
throw std::exception("No video stream in input video file.");
// find decoder using id
codec = avcodec_find_decoder(formatCtx->streams[streamIndex]->codec->codec_id);
if (codec == nullptr)
throw std::exception("Could not find suitable decoder for input file.");
// copy context from input stream
codecCtx = avcodec_alloc_context3(codec);
if (avcodec_copy_context(codecCtx, formatCtx->streams[streamIndex]->codec) != 0)
throw std::exception("Could not copy codec context from input stream.");
if (avcodec_open2(codecCtx, codec, NULL) < 0)
throw std::exception("Could not open decoder.");
}
// frame must be initialized with av_frame_alloc() before!
// Returns true if there are other frames, false if not.
// success == 1 if frame is valid, 0 if not.
bool InputVideoHandler::readFrame(AVFrame* frame, int* success) {
*success = 0;
if (av_read_frame(formatCtx, &packet) < 0)
return false;
if (packet.stream_index == streamIndex) {
avcodec_decode_video2(codecCtx, frame, success, &packet);
}
av_free_packet(&packet);
return true;
}
// This class opens the output and write frames to it
class OutputVideoBuilder{
public:
OutputVideoBuilder(char* name, AVCodecContext* inputCtx);
~OutputVideoBuilder();
void writeFrame(AVFrame* frame);
void writeVideo();
private:
OutputVideoBuilder();
void init(char* name, AVCodecContext* inputCtx);
void logMsg(AVPacket* packet, AVRational* tb);
AVFormatContext* formatCtx;
AVCodec* codec;
AVCodecContext* codecCtx;
AVStream* stream;
};
void OutputVideoBuilder::init(char* name, AVCodecContext* inputCtx) {
if (avformat_alloc_output_context2(&formatCtx, NULL, NULL, name) < 0)
throw std::exception("Could not determine file extension from provided name.");
codec = avcodec_find_encoder(inputCtx->codec_id);
if (codec == nullptr) {
throw std::exception("Could not find suitable encoder.");
}
codecCtx = avcodec_alloc_context3(codec);
if (avcodec_copy_context(codecCtx, inputCtx) < 0)
throw std::exception("Could not copy output codec context from input");
codecCtx->time_base = inputCtx->time_base;
codecCtx->compression_level = 0;
if (avcodec_open2(codecCtx, codec, NULL) < 0)
throw std::exception("Could not open encoder.");
stream = avformat_new_stream(formatCtx, codec);
if (stream == nullptr) {
throw std::exception("Could not allocate stream.");
}
stream->id = formatCtx->nb_streams - 1;
stream->codec = codecCtx;
stream->time_base = codecCtx->time_base;
av_dump_format(formatCtx, 0, name, 1);
if (!(formatCtx->oformat->flags & AVFMT_NOFILE)) {
if (avio_open(&formatCtx->pb, name, AVIO_FLAG_WRITE) < 0) {
throw std::exception("Could not open output file.");
}
}
if (avformat_write_header(formatCtx, NULL) < 0) {
throw std::exception("Error occurred when opening output file.");
}
}
void OutputVideoBuilder::writeFrame(AVFrame* frame) {
AVPacket packet = { 0 };
int success;
av_init_packet(&packet);
if (avcodec_encode_video2(codecCtx, &packet, frame, &success))
throw std::exception("Error encoding frames");
if (success) {
av_packet_rescale_ts(&packet, codecCtx->time_base, stream->time_base);
packet.stream_index = stream->index;
logMsg(&packet,&stream->time_base);
av_interleaved_write_frame(formatCtx, &packet);
}
av_free_packet(&packet);
}
This is the part of the main function that reads and write frames:
while (inputHandler->readFrame(frame,&gotFrame)) {
if (gotFrame) {
try {
outputBuilder->writeFrame(frame);
}
catch (std::exception e) {
std::cout << e.what() << std::endl;
return -1;
}
}
}
Your qmin/qmax answer is partially correct, but it misses the point, in that the quality indeed goes up, but the compression ratio (in terms of quality per bit) will suffer significantly as you restrict the qmin/qmax range - i.e. you will spend many more bits to accomplish the same quality than should really be necessary if you used the encoder optimally.
To increase quality without hurting the compression ratio, you need to actually increase the quality target. How you do this differs a little depending on the codec, but you typically increase the target CRF value or target bitrate. For commandline options, see e.g. the H264 docs. There's identical docs for HEVC/VP9 also. To use these options in the C API, use av_opt_set() with the same option names/values.
In case this could be useful to someone else, I add the answer that damjeux suggested, which worked for me. AVCodecContex has two members qmin and qmax which control the QP (quantization parameter) of the encoder. By default in my case qmin is 2 and qmax is 31. By setting qmax to a lower value the quality of the output improves.
I am trying to have a play sound method execute on a click event, followed by a stop method being called on the release, using OpenAL in in C++. My problem is that I cannot get it to stop playing on the release. My source code to play the sound is as follows:
bool SoundManager::play(QString fileName, float pitch, float gain)
{
static uint sourceIndex = 0;
ALint state;
// Get the corresponding buffer id set up in the init function.
ALuint bufferID = mSoundBuffers[fileName];
if (bufferID != 0) {
// Increment which source we are using, so that we play in a "free" source.
sourceIndex = (sourceIndex + 1) % SOUNDMANAGER_MAX_NBR_OF_SOURCES;
// Get the source in which the sound will be played.
ALuint source = mSoundSources[sourceIndex];
if (alIsSource (source) == AL_TRUE) {
// Attach the buffer to an available source.
alSourcei(source, AL_BUFFER, bufferID);
if (alGetError() != AL_NO_ERROR) {
reportOpenALError();
return false;
}
// Set the source pitch value.
alSourcef(source, AL_PITCH, pitch);
if (alGetError() != AL_NO_ERROR) {
reportOpenALError();
return false;
}
// Set the source gain value.
alSourcef(source, AL_GAIN, gain);
if (alGetError() != AL_NO_ERROR) {
reportOpenALError();
return false;
}
alGetSourcei(source, AL_SOURCE_STATE, &state);
if (state!=AL_PLAYING)
alSourcePlay(source);
else if(state==AL_PLAYING)
alSourceStop(source);
if (alGetError() != AL_NO_ERROR) {
reportOpenALError();
return false;
}
}
} else {
// The buffer was not found.
return false;
}`
I think that the issue is that when it is called the second time, when it should be stopped, it is a different source, and that is why its state is not playing. If this is the issue, then how can I access the same source?
Of course it's not the same source as before, you increase the sourceIndex variable each call.
So the first call, to play, sourceIndex will be 1 (sourceIndex + 1). The next time you call the function (which btw. is badly named for something that toggles playing) then sourceIndex again will be increased by one, which will give you a new index into the source vector.