I wrote the program that takes as an input some .opus file, decodes it using libavcodec and libopus and then plays it using SDL2. Program works on Windows 10 and Ubuntu 18.04, however it crashes with the segmentation fault on Debian 9.
I've tried to update libavcodec and libopus libraries, tried to compile using clang and gcc - nothing helped.
Address sanitizer shows that stack overflow happens:
ASAN:DEADLYSIGNAL
=================================================================
==12167==ERROR: AddressSanitizer: stack-overflow on address 0x2b3e74c81ff8 (pc 0x2b3e7a098803 bp 0x2b3e74c82690 sp 0x2b3e74c81eb0 T2)
#0 0x2b3e7a098802 in quant_all_bands celt/bands.c:1403
#1 0x2b3e7a0a2a37 in celt_decode_with_ec celt/celt_decoder.c:1083
#2 0x2b3e7a0c8afb in opus_decode_frame src/opus_decoder.c:518
#3 0x2b3e7a0c9e40 in opus_decode_native src/opus_decoder.c:721
#4 0x2b3e7a0d33f3 in opus_multistream_decode_native src/opus_multistream_decoder.c:253
#5 0x2b3e7a0d37a8 in opus_multistream_decode src/opus_multistream_decoder.c:398
#6 0x2b3e760ad83c (/usr/lib/x86_64-linux-gnu/libavcodec.so.57+0x43583c)
#7 0x2b3e75e4ca27 (/usr/lib/x86_64-linux-gnu/libavcodec.so.57+0x1d4a27)
#8 0x2b3e75e4f62a in avcodec_send_packet (/usr/lib/x86_64-linux-gnu/libavcodec.so.57+0x1d762a)
#9 0x2b3e75e4f9e6 (/usr/lib/x86_64-linux-gnu/libavcodec.so.57+0x1d79e6)
#10 0x55ef09511882 in decode(AVCodecContext*, AVPacket*, unsigned char*, int) /home/ram/my/player3/speaker.cpp:296
#11 0x55ef09511626 in fillBuffer(AVCodecContext*, unsigned char*, int) /home/ram/my/player3/speaker.cpp:251
#12 0x55ef09511294 in process(AVCodecContext*, unsigned char*, int) /home/ram/my/player3/speaker.cpp:194
#13 0x55ef095105b9 in audio_callback(void*, unsigned char*, int) /home/ram/my/player3/speaker.cpp:69
#14 0x2b3e7815cc31 (/usr/lib/x86_64-linux-gnu/libSDL2-2.0.so.0+0x1fc31)
#15 0x2b3e781bcf8b (/usr/lib/x86_64-linux-gnu/libSDL2-2.0.so.0+0x7ff8b)
#16 0x2b3e7820c6c8 (/usr/lib/x86_64-linux-gnu/libSDL2-2.0.so.0+0xcf6c8)
#17 0x2b3e77be74a3 in start_thread (/lib/x86_64-linux-gnu/libpthread.so.0+0x74a3)
#18 0x2b3e7940ed0e in __clone (/lib/x86_64-linux-gnu/libc.so.6+0xe8d0e)
SUMMARY: AddressSanitizer: stack-overflow celt/bands.c:1403 in quant_all_bands
Thread T2 (SDLAudioDev2) created by T0 here:
#0 0x2b3e74d0df59 in __interceptor_pthread_create (/usr/lib/x86_64-linux-gnu/libasan.so.3+0x30f59)
#1 0x2b3e7820c732 (/usr/lib/x86_64-linux-gnu/libSDL2-2.0.so.0+0xcf732)
I also tried to increase stack size using ulimit -s unlimited and tried to increase stack size for the thread that starts decoding, didn't work.
In main.cpp file I have this:
#include <iostream>
#include <memory>
#include <mutex>
#include "speaker.h"
#include "SDL2/SDL.h"
extern "C"{
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
}
static int decode_audio_file(const char* path) {
av_register_all();
// get format from audio file
AVFormatContext* format = avformat_alloc_context();
if (avformat_open_input(&format, path, NULL, NULL) != 0) {
std::cout << "Could not open file" << std::endl;
return -1;
}
if (avformat_find_stream_info(format, NULL) < 0) {
std::cout << "Could not retrieve stream info from file" << std::endl;
return -1;
}
// Find the index of the first audio stream
int stream_index =- 1;
for (int i=0; i< format->nb_streams; i++) {
if (format->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
stream_index = i;
break;
}
}
if (stream_index == -1) {
std::cout << "Could not retrieve audio stream from file" << std::endl;
return -1;
}
AVStream* stream = format->streams[stream_index];
// Initialize speaker
init_Speaker("OPUS",
48000,
2,
15,
3,
av_get_channel_layout("stereo"),
av_get_sample_fmt("s16"));
// prepare to read data
AVPacket* packet;
packet = av_packet_alloc();
av_init_packet(packet);
// iterate through frames
while (av_read_frame(format, packet) >= 0) {
play(packet->data, packet->size, std::chrono::microseconds{packet->pts},
std::chrono::microseconds{packet->dts});
av_packet_unref(packet);
}
// clean up
avformat_free_context(format);
close_Speaker();
// success
return 0;
}
int main(int argc, char const *argv[]) {
// check parameters
if (argc < 2) {
std::cout << "Please provide the path to an audio file as first command-line argument.\n";
return -1;
}
// Init Audio
SDL_Init(SDL_INIT_AUDIO);
// decode data
if (decode_audio_file(argv[1]) != 0) {
return -1;
}
std::cout << "Finish" << std::endl;
return 0;
}
In speaker.cpp:
#include "speaker.h"
#include "pthread.h"
#include "avcodec.h"
#include "common.h"
#include <iostream>
extern "C"
{
#include <libswresample/swresample.h>
#include <libavutil/hwcontext.h>
}
using std::chrono::microseconds;
SDL_AudioDeviceID m_id;
AVCodecParserContext *parser = nullptr;
//
constexpr static auto buffer_size{1024}; // 2048
constexpr static auto buffer_max_size{AVCODEC_MAX_AUDIO_FRAME_SIZE * 4};
uint32_t m_samplerate;
uint32_t m_queue_limit;
uint32_t m_queue_dropfactor;
int64_t m_channel_layout;
AVSampleFormat m_device_format;
AVCodecID audio_codec_id{AV_CODEC_ID_NONE};
AVCodecContext* adecoder;
player::PacketQueue queue(0, true);
static uint8_t* buf = nullptr;
uint32_t absize{};
uint32_t abpos{};
int32_t max_decoder_size{};
// need a converter?
uint8_t* convbuf{};
SwrContext* swrctx{};
int32_t sframes{};
AVCodec* codec{};
uint8_t * audio_buffer_init() {
if(buf == nullptr) {
buf = (uint8_t*) malloc(buffer_max_size);
if(buf == nullptr) {
return nullptr;
}
}
return buf;
}
void audio_callback(void* userdata, uint8_t* stream, int len) {
AVCodecContext* decoder = (AVCodecContext*)userdata;
process(decoder, stream, len);
};
void init_Speaker(const std::string& codecName,
int32_t samplerate,
uint8_t channels,
uint32_t queue_limit,
uint32_t queue_dropfactor,
int64_t channel_layout,
AVSampleFormat format)
{
m_samplerate = samplerate;
m_queue_limit = queue_limit;
m_queue_dropfactor = queue_dropfactor;
m_device_format = format;
m_channel_layout = channel_layout;
SDL_SetHint(SDL_HINT_THREAD_STACK_SIZE, "8388608");
if(codecName.empty())
throw std::runtime_error("audio decoder: no codec specified.");
auto names = player::lookup_ffmpeg_decoders(codecName);
if(names == nullptr)
throw std::runtime_error("audio decoder: cannot find decoder names for {}"+codecName);
audio_codec_id = player::lookup_codec_id(codecName);
codec = player::avcodec_find_decoder(names, AV_CODEC_ID_NONE);
if(codec == nullptr)
throw std::runtime_error("audio decoder: cannot find the decoder for {}"+codecName);
adecoder = avcodec_alloc_context3(codec);
if(adecoder == nullptr)
throw std::runtime_error("audio decoder: cannot allocate context");
adecoder->channels = channels;
adecoder->sample_rate = samplerate;
if(adecoder->channels == 1)
{
adecoder->channel_layout = AV_CH_LAYOUT_MONO;
}
else if(adecoder->channels == 2)
{
adecoder->channel_layout = AV_CH_LAYOUT_STEREO;
}
else
throw std::runtime_error("audio decoder: unsupported number of channels ({})"+ adecoder->channels);
if(avcodec_open2(adecoder, codec, nullptr) != 0)
throw std::runtime_error("audio decoder: cannot open decoder");
parser = av_parser_init(codec->id);
SDL_AudioSpec wanted, spec;
wanted.freq = samplerate;
wanted.format = AUDIO_S16SYS;
wanted.channels = channels;
wanted.silence = 0;
wanted.samples = buffer_size;
wanted.userdata = adecoder;
wanted.callback = audio_callback;
m_id = SDL_OpenAudioDevice(nullptr, 0, &wanted, &spec, 0);
if(m_id == 0)
throw std::runtime_error(SDL_GetError());
SDL_PauseAudioDevice(m_id, 0);
}
void close_Speaker()
{
SDL_CloseAudioDevice(m_id);
if(adecoder != nullptr)
player::avcodec_close(adecoder);
}
void play(uint8_t* buffer, size_t bufsize, microseconds pts, microseconds dts)
{
if(!buffer || !bufsize) {
return;
}
AVPacket* avpkt;
avpkt = av_packet_alloc();
av_init_packet(avpkt);
uint8_t bf[bufsize + 64];
memcpy(bf, buffer, bufsize);
av_parser_parse2(parser, adecoder, &avpkt->data, &avpkt->size,
bf, bufsize,
pts.count(), dts.count(), 0);
queue.put(av_packet_clone(avpkt));
queue.drop(m_queue_limit, m_queue_dropfactor);
}
//
void process(AVCodecContext* decoder, uint8_t* stream, int ssize)
{
auto filled = fillBuffer(decoder, stream, ssize);
auto unfilled{(ssize - filled) / 4};
auto dummy{sframes};
sframes = unfilled == 0 ? 0 : sframes + unfilled;
memset(stream + filled, 0, unfilled * 4);
if(sframes != dummy)
queue.add_silence((int64_t)sframes * 1000000 / m_samplerate);
}
int fillBuffer(AVCodecContext* decoder, uint8_t* stream, int ssize)
{
int filled{};
AVPacket avpkt;
audio_buffer_init();
while(filled < ssize)
{
int dsize{}, delta{};
// buffer has enough data
if(absize - abpos >= static_cast<unsigned int>(ssize - filled))
{
delta = ssize - filled;
std::copy(buf + abpos, buf + abpos + delta, stream);
abpos += delta;
filled += delta;
return ssize;
}
else if(absize - abpos > 0)
{
delta = absize - abpos;
std::copy(buf + abpos, buf + abpos + delta, stream);
stream += delta;
filled += delta;
abpos = absize = 0;
}
// move data to head, leave more ab buffers
if(abpos != 0)
{
std::copy(buf + abpos, buf + abpos + absize - abpos, buf);
absize -= abpos;
abpos = 0;
}
// decode more packets
if(!queue.get(&avpkt, false))
break;
if((dsize = decode(decoder, &avpkt, buf + absize, buffer_max_size - absize)) < 0)
break;
absize += dsize;
}
return filled;
}
int decode(AVCodecContext* decoder, AVPacket* pkt, uint8_t* dstbuf, int dstlen)
{
const uint8_t* srcplanes[SWR_CH_MAX];
uint8_t* dstplanes[SWR_CH_MAX];
int filled{};
AVFrame* aframe = av_frame_alloc();
auto saveptr = pkt->data;
while(pkt->size > 0)
{
int len{}, got_frame{};
unsigned char* srcbuf{};
int datalen{};
if((len = avcodec_decode_audio4(decoder, aframe, &got_frame, pkt)) < 0)
{
return -1;
}
if(got_frame == 0)
{
pkt->size -= len;
pkt->data += len;
continue;
}
if(aframe->format == m_device_format)
{
datalen = av_samples_get_buffer_size(nullptr,
aframe->channels /*rtspconf->audio_channels*/,
aframe->nb_samples,
(AVSampleFormat)aframe->format,
1 /*no-alignment*/);
srcbuf = aframe->data[0];
}
else
{
// need conversion!
if(swrctx == nullptr)
{
if((swrctx = swr_alloc_set_opts(nullptr,
m_channel_layout,
m_device_format,
m_samplerate,
aframe->channel_layout,
(AVSampleFormat)aframe->format,
aframe->sample_rate,
0,
nullptr)) == nullptr)
{
return -1;
}
auto err = swr_init(swrctx);
if(err < 0)
{
char msg[1024];
av_strerror(err, msg, 1024);
return -1;
}
max_decoder_size = av_samples_get_buffer_size(nullptr,
2,
m_samplerate,
m_device_format,
1 /*no-alignment*/);
if((convbuf = (unsigned char*)::malloc(max_decoder_size)) == nullptr)
{
return -1;
}
}
datalen = av_samples_get_buffer_size(nullptr,
2,
aframe->nb_samples,
m_device_format,
1 /*no-alignment*/);
if(datalen > max_decoder_size)
{
return -1;
}
srcplanes[0] = aframe->data[0];
if(av_sample_fmt_is_planar((AVSampleFormat)aframe->format) != 0)
{
// planar
int i;
for(i = 1; i < aframe->channels; i++)
{
srcplanes[i] = aframe->data[i];
}
srcplanes[i] = nullptr;
}
else
{
srcplanes[1] = nullptr;
}
dstplanes[0] = convbuf;
dstplanes[1] = nullptr;
swr_convert(swrctx, dstplanes, aframe->nb_samples, srcplanes, aframe->nb_samples);
srcbuf = convbuf;
}
if(datalen > dstlen)
{
datalen = dstlen;
}
std::copy(srcbuf, srcbuf + datalen, dstbuf);
dstbuf += datalen;
dstlen -= datalen;
filled += datalen;
pkt->size -= len;
pkt->data += len;
av_frame_unref(aframe);
}
pkt->data = saveptr;
if(pkt->data)
av_packet_unref(pkt);
if(aframe != nullptr)
av_frame_free(&aframe);
return filled;
}
In packet_queue.cpp:
#include "packet_queue.h"
using player::PacketQueue;
using lock_guard = std::lock_guard<std::mutex>;
using unique_lock = std::unique_lock<std::mutex>;
using std::chrono::milliseconds;
PacketQueue::PacketQueue(uint32_t playback_queue_silence, bool playback_queue_debug) :
m_playback_queue_debug(playback_queue_debug), m_playback_queue_silence(playback_queue_silence)
{
}
void PacketQueue::clear()
{
lock_guard lk{m_mtx};
for(auto& pkt : queue)
av_packet_unref(pkt);
m_size = 0;
queue.clear();
}
void PacketQueue::add_silence(int64_t silence_pts)
{
if(m_playback_queue_silence == 0)
{
return;
}
lock_guard lk{m_mtx};
silence_pts = filtered_packets > 0 ? silence_pts : last_pts + silence_pts;
auto tv = std::chrono::microseconds{last_pts};
auto tv2 = std::chrono::microseconds{silence_pts};
}
bool PacketQueue::put(AVPacket* pkt)
{
if(pkt == nullptr)
{
return false;
}
lock_guard lk{m_mtx};
if((silence_pts - pkt->pts) > (m_playback_queue_silence * 1000))
{
auto tv = std::chrono::microseconds{pkt->pts};
filtered_packets++;
if(m_playback_queue_debug)
return true;
}
queue.push_back(pkt);
filtered_packets = 0;
return true;
}
bool PacketQueue::get(AVPacket* pkt, bool block, milliseconds timeout)
{
unique_lock lk{m_mtx};
for(;;)
{
if(queue.size() > 0)
{
auto ptr = queue.front();
queue.pop_front();
m_size -= ptr->size;
last_pts = ptr->pts;
av_packet_move_ref(pkt, ptr);
return true;
}
else if(!block)
{
return false;
}
else if(!m_cv.wait_for(lk, timeout, [&] { return !queue.empty(); }))
{
return false;
}
}
return false;
}
bool PacketQueue::drop(size_t limit, size_t dropfactor)
{
int dropped, count = 0;
lock_guard lk{m_mtx};
// queue size exceeded?
if(queue.size() <= limit)
{
return false;
}
// start dropping
dropped = queue.size() / dropfactor;
// keep at least one
if(dropped == queue.size())
dropped--;
AVPacket* pkt;
while(dropped-- > 0 && !queue.empty())
{
pkt = queue.front();
if(pkt->flags != AV_PKT_FLAG_KEY)
{
queue.pop_front();
m_size -= pkt->size;
av_packet_unref(pkt);
++count;
}
}
return true; // count;
}
int PacketQueue::drop2(size_t limit, bool error)
{
int count = 0;
// dropping enabled?
if(limit <= 0 && !error)
return 0;
lock_guard lk{m_mtx};
// queue size exceeded?
if(queue.size() <= limit && !error)
return false;
for(auto i = queue.begin(); i != queue.end();)
{
AVPacket* pkt = *i;
if(pkt->flags != AV_PKT_FLAG_KEY)
{
m_size -= pkt->size;
av_packet_unref(pkt);
i = queue.erase(i);
count++;
}
else
++i;
}
return count;
}
1403 line of celt/bands.c: screenshot
Versions of libraries that I tried on Debian 9: libavcodec.so.57.64.101 and libopus.so.0.5.3
also I built manually libavcodec.so.57.107.100 and libopus.so.0.8.0 and tried to use them - the same error appears.
As I already mentioned, everything works fine on Windows 10 and Ubuntu 18.04. So I have no clue what could be the reason of the issue. Any help is appreciated.