My preset:
m_stEncodeConfig.encodeCodecConfig.hevcConfig.sliceMode = 3u;
m_stEncodeConfig.encodeCodecConfig.hevcConfig.sliceModeData = (uint32_t)m_stEncodeStreamInfo.nMaxSliceNum; //4
m_stCreateEncodeParams.reportSliceOffsets = 1;
m_stCreateEncodeParams.enableSubFrameWrite = 1;
code of process output:
NV_ENC_LOCK_BITSTREAM lockBitstreamData;
memset(&lockBitstreamData, 0, sizeof(lockBitstreamData));
lockBitstreamData.version = NV_ENC_LOCK_BITSTREAM_VER;
lockBitstreamData.outputBitstream = pEncodeBuffer->stOutputBfr.hBitstreamBuffer;
lockBitstreamData.doNotWait = 1u;
std::vector<uint32_t> arrSliceOffset(m_stEncodeConfig.encodeCodecConfig.hevcConfig.sliceModeData);
lockBitstreamData.sliceOffsets = arrSliceOffset.data();
while (true)
{
NVENCSTATUS status = m_pEncodeAPI->nvEncLockBitstream(m_hEncoder, &lockBitstreamData);
auto tick = int(std::chrono::steady_clock::now().time_since_epoch().count() / 1000000);
if (status == NVENCSTATUS::NV_ENC_SUCCESS)
{
if (lockBitstreamData.hwEncodeStatus == 2)
{
static std::ofstream of("slice.h265", std::ios::trunc | std::ios::binary);
of.write((char*)lockBitstreamData.bitstreamBufferPtr, lockBitstreamData.bitstreamSizeInBytes);
of.flush();
break;
}
NVENCAPI_CALL_CHECK(m_pEncodeAPI->nvEncUnlockBitstream(m_hEncoder, lockBitstreamData.outputBitstream));
}
else
{
break;
}
}
play bitstream:
ffplay -i slice.h265
output : Packet corrupt
arrSliceOffset[0] always = 255.
I watch the memory from VS Debug and compare with enableSubFrameWrite = 0,the bitstreamSizeInBytes less of valid data size . It’s BUG or I loss some details? Anybady can tell me how can I correct use of enableSubFrameWrite
Related
NVIDIA Gefore GTX 1660Ti graphics card, I use video-sdk-samples for HEVC encoding, YUV format: NV_ENC_BUFFER_FORMAT_YUV444, resolution “1920 * 1080” and “3840 * 2160” resolution encoding is successful, but resolution 2560*1440 encoding, prompting error info :NV_ENC_ERR_INVALID_PARAM, I do not know where the problem lies?
void NvEncoder::DoEncode(NV_ENC_INPUT_PTR inputBuffer, std::vector<bool> &vKeyFrame, std::vector<std::vector<uint8_t>> &vPacket, NV_ENC_PIC_PARAMS *pPicParams)
{
NV_ENC_PIC_PARAMS picParams = {};
if (pPicParams)
{
picParams = *pPicParams;
}
picParams.version = NV_ENC_PIC_PARAMS_VER;
picParams.pictureStruct = NV_ENC_PIC_STRUCT_FRAME;
picParams.inputBuffer = inputBuffer;
picParams.bufferFmt = GetPixelFormat();
picParams.inputWidth = GetEncodeWidth();
picParams.inputHeight = GetEncodeHeight();
picParams.outputBitstream = m_vBitstreamOutputBuffer[m_iToSend % m_nEncoderBuffer];
picParams.completionEvent = m_vpCompletionEvent[m_iToSend % m_nEncoderBuffer];
NVENCSTATUS nvStatus = m_nvenc.nvEncEncodePicture(m_hEncoder, &picParams);
if (nvStatus == NV_ENC_SUCCESS || nvStatus == NV_ENC_ERR_NEED_MORE_INPUT)
{
m_iToSend++;
GetEncodedPacket(m_vBitstreamOutputBuffer, vKeyFrame, vPacket, false);
}
else
{
printf("picParams.inputWidth = %d,picParams.inputHeight = %d \n", picParams.inputWidth, picParams.inputHeight);
NVENC_THROW_ERROR("nvEncEncodePicture API failed", nvStatus);
}
}
error:
nvEncEncodePicture Prompt Error:NV_ENC_ERR_INVALID_PARAM
`
1. About the problem
I'm trying to send data store in a array called UxbTx[64] to 2 Hid devices that have the same VendorID(VID) and ProductID(PID) but differ in SerialNum (this differnce help me to enumerate 2 devices) and connect to PC via Usb Hub . I have successfully recognized two device and use hid_write() to send data but just 1 device acting right with my data config. For example, i want to turn on 2 devices but device 1 ON whereas the device 2 still remain OFF.
2. What I have tried
At first I thought that I have failed to send to the second device but it not. I use res=hid_write() and it return 0 which means successfully send for both devices.
This is the code that i use:
static hid_device** id_device;
static int max_size;
int res;
struct device_info
{
wchar_t* serial_num;
int id;
};
std::vector<device_info> devEnum(unsigned short vendor_id, unsigned short product_id)
{
std::vector<device_info> device;
int count = 0, total = 0, res = 0,i=0;
hid_device_info* dev = NULL, * cur_dev = NULL;
hid_device* temp_handle = NULL;
std::vector<wchar_t*> string;
wchar_t wstr[MAX_STR], temp[MAX_STR];
device_info inf;
hid_enumerate(vendor_id, product_id);
res = hid_init();
dev = hid_enumerate(0x461, 0x20);
for (cur_dev = dev; cur_dev != NULL; cur_dev = cur_dev->next)
{
memcpy(temp, cur_dev->serial_number, MAX_STR);
wcsncpy(temp, cur_dev->serial_number, MAX_STR);
temp[MAX_STR - 1] = L'\0';
string.push_back(temp);
inf.serial_num = cur_dev->serial_number;
inf.id = count;
device.push_back(inf);
count++;
}
max_size = device.size();
return device;
}
int NhgIsOpen(wchar_t* Manufacturer, wchar_t* Product, wchar_t* SerialNumber, std::vector<device_info> devices) {
int length = devices.size();
int res = 0;
id_device = new hid_device * [length];
wchar_t KeyManufacturer[MAX_STR] = L"test";
wchar_t KeyProduct[MAX_STR] = L"device test";
for (int i = 0; i < length; i++)
{
id_device[i] = hid_open(0x123, 0x15, devices[i].serial_num);
if (!id_device[i])
return -2;
hid_set_nonblocking(id_device[i], 0);
res = hid_get_manufacturer_string(id_device[i], Manufacturer, MAX_STR);
if (wcscmp(Manufacturer, KeyManufacturer) != 0)
return -3; //Manufacturer not match
res = hid_get_product_string(id_device[i], Product, MAX_STR);
if (wcscmp(Product, KeyProduct) != 0)
return -4; // KeyProdeuct not match
}
return 0;
}
INT32 IoControl(UINT16 IoState, int axis_id) {
UINT8 UsbTx[64];
//clear TX buffer
std::fill_n(UsbTx, 64, 0);
//report byte
UsbTx[0] = 0x01;
//USB user define cmd
UsbTx[1] = 0x00;
UsbTx[2] = 0x15; // turn on device
UsbTx[11] = 0x00;
UsbTx[12] = (0xFF) & IoState;
res = hid_write(id_device[axis_id], UsbTx, 64);
if (res < 0)
return -5; //can't write
return 0;
}
3. Question
If i can enumerate 2 devices does it means i can talk to 2 devices simultaneously?
Does the Report ID in firmware is the problem? I mean does i have to re-config Report ID in the Descriptor of the device so they can read the data at the same time?
I trying get the devices from pjsua2 , I got it get all devices, but do not got split in capture device and playback device.
void AudioController::load(){
Endpoint ep;
ep.libCreate();
// Initialize endpoint
EpConfig ep_cfg;
ep.libInit( ep_cfg );
AudDevManager &manager = ep.audDevManager();
manager.refreshDevs();
this->input.clear();
const AudioDevInfoVector &list = manager.enumDev();
for(unsigned int i = 0;list.size() != i;i++){
AudioDevInfo * info = list[i];
GtAudioDevice * a = new GtAudioDevice();
a->name = info->name.c_str();
a->deviceId = i;
qDebug() << info->name.c_str();
qDebug() << info->driver.c_str();
qDebug() << info->caps;
this->input.append(a);
}
ep.libDestroy();
}
This is my output:
Wave mapper
WMME
23
Microfone (Dispositivo de High
WMME
3
Alto-falantes (Dispositivo de H
WMME
21
You can check the fields inputCount and outputCount inside AudioDevInfo.
According the documentation:
unsigned inputCount
Maximum number of input channels supported by this device. If the
value is zero, the device does not support input operation (i.e. it is
a playback only device).
And
unsigned outputCount
Maximum number of output channels supported by this device. If the
value is zero, the device does not support output operation (i.e. it
is an input only device).
So you could do something like this:
for(unsigned int i = 0;list.size() != i;i++){
AudioDevInfo * info = list[i];
GtAudioDevice * a = new GtAudioDevice();
a->name = info->name.c_str();
a->deviceId = i;
if (info->inputCount > 0) {
a->captureDevice = true;
}
if (info->outputCount > 0) {
a->playbackDevice = true;
}
this->input.append(a);
}
Reference: http://www.pjsip.org/pjsip/docs/html/structpj_1_1AudioDevInfo.htm
Another way, you can check the field caps (capabilities). Something like this:
for (int i = 0; i < list.size(); i++)
{
AudioDevInfo * info = list[i];
if ((info.caps & (int)pjmedia_aud_dev_cap.PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY) != 0)
{
// Playback devices come here
}
if ((info.caps & (int)pjmedia_aud_dev_cap.PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY) != 0)
{
// Capture devices come here
}
}
caps is combined from these possible values:
enum pjmedia_aud_dev_cap {
PJMEDIA_AUD_DEV_CAP_EXT_FORMAT = 1,
PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY = 2,
PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY = 4,
PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING = 8,
PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING = 16,
PJMEDIA_AUD_DEV_CAP_INPUT_SIGNAL_METER = 32,
PJMEDIA_AUD_DEV_CAP_OUTPUT_SIGNAL_METER = 64,
PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE = 128,
PJMEDIA_AUD_DEV_CAP_INPUT_SOURCE = 128,
PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE = 256,
PJMEDIA_AUD_DEV_CAP_EC = 512,
PJMEDIA_AUD_DEV_CAP_EC_TAIL = 1024,
PJMEDIA_AUD_DEV_CAP_VAD = 2048,
PJMEDIA_AUD_DEV_CAP_CNG = 4096,
PJMEDIA_AUD_DEV_CAP_PLC = 8192,
PJMEDIA_AUD_DEV_CAP_MAX = 16384
}
i using this code to encode video stream using vp8 and i decided to give vp9 a try so i changed every thing with starts with vp_* from 8 to 9.
but the vp9 encoder always return a null packet although the encoder doesn't return any error.
here is the code i'am using for configuring.
vpx_codec_err_t error = vpx_codec_enc_config_default(vpx_codec_vp9_cx(), &enc_cfg, 0);
if(error != VPX_CODEC_OK)
return error;
enc_cfg.g_timebase.den = fps;
enc_cfg.rc_undershoot_pct = 95;
enc_cfg.rc_target_bitrate = bitrate;
enc_cfg.g_error_resilient = 1;
enc_cfg.kf_max_dist = 999999;
enc_cfg.rc_buf_initial_sz = 4000;
enc_cfg.rc_buf_sz = 6000;
enc_cfg.rc_buf_optimal_sz = 5000;
enc_cfg.rc_end_usage = VPX_CBR;
enc_cfg.g_h = height;
enc_cfg.g_w = width;
enc_cfg.rc_min_quantizer = 4;
enc_cfg.rc_max_quantizer = 56;
enc_cfg.g_threads = 4;
enc_cfg.g_pass = VPX_RC_ONE_PASS;
error = vpx_codec_enc_init(&codec, vpx_codec_vp9_cx(), &enc_cfg, 0);
if(error != VPX_CODEC_OK)
return error;
vpx_img_alloc(&vpx_image,VPX_IMG_FMT_I420 , width, height, 1);
configured = true;
return VPX_CODEC_OK;
and the code for the encoding
libyuv::RAWToI420(frame, vpx_image.d_w * 3, vpx_image.planes[VPX_PLANE_Y],vpx_image.stride[VPX_PLANE_Y],
vpx_image.planes[VPX_PLANE_U], vpx_image.stride[VPX_PLANE_U], vpx_image.planes[VPX_PLANE_V],
vpx_image.stride[VPX_PLANE_V], vpx_image.d_w, vpx_image.d_h);
const vpx_codec_cx_pkt_t *pkt;
vpx_codec_err_t error = vpx_codec_encode(&codec, &vpx_image, 0, 1, 0, VPX_DL_GOOD_QUALITY);
if(error != VPX_CODEC_OK)
return vector<byte>();
vpx_codec_iter_t iter = NULL;
if((pkt = vpx_codec_get_cx_data(&codec, &iter)))//always return null ?
{
if(pkt->kind == VPX_CODEC_CX_FRAME_PKT)
{
int length = pkt->data.frame.sz;
byte* buf = (byte*) pkt->data.frame.buf;
vector<byte> data(buf, buf + length);
return data;
}
return vector<byte>();
}
return vector<byte>();
the code is fully working if i'am using vp8 instead of 9, any help is welcomed
Just came across this post because I faced the same problem. Just for other to know: I solved it with setting
enc_cfg.g_lag_in_frames = 0;
This basically disallows the encoder to consume up to default 25 frames until it produces any output.
I write an application, which plays a sound getting from Hardware (like a ring buffer filled with a sinus wave with certain frequency). Everything works fine, and I can playback the created sound correctly except a periodical clicking (maybe at the end of buffer?) and noise.
I initialize and run the Buffer:
void Audiooutput::InitializeAudioParameters()
{
Audio_DataWritten = 0;
Audio_fragments = 4;
Audio_channels = 2;
Audio_BufferSize = 256;
Audio_Samplerate = 8000;
Audio_ResamplingFactor = 1;
Audio_Framesize = 2;
// (SND_PCM_FORMAT_S16_LE / 8);
Audio_frames = Audio_BufferSize / Audio_Framesize * Audio_fragments;
snd_pcm_uframes_t size;
err = snd_pcm_hw_params_any(pcmPlaybackHandle, hw_params);
err = snd_pcm_hw_params_set_rate_resample(pcmPlaybackHandle, hw_params, 1);
// qDebug()<<a1.sprintf(" % d \t snd_pcm_hw_params_set_rate: %s",Audio_Samplerate,snd_strerror(err));
err =
snd_pcm_hw_params_set_format(pcmPlaybackHandle, hw_params,
SND_PCM_FORMAT_S16_LE);
err =
snd_pcm_hw_params_set_channels(pcmPlaybackHandle, hw_params,
Audio_channels);
err = snd_pcm_hw_params_set_rate_near(pcmPlaybackHandle, hw_params, &Audio_Samplerate, 0);
// qDebug()<<a1.sprintf(" % d \t snd_pcm_hw_params_set_rate: %s",Audio_Samplerate,snd_strerror(err));
if ((err =
snd_pcm_hw_params_set_periods_near(pcmPlaybackHandle, hw_params,
&Audio_fragments, 0)) < 0) {
qDebug() << a1.sprintf("Error setting # fragments to %d: %s\n",
Audio_fragments, snd_strerror(err));
} else
qDebug() << a1.sprintf("setting # fragments to %d: %s\n",
Audio_fragments, snd_strerror(err));
err = snd_pcm_hw_params_get_buffer_size(hw_params, &size);
if ((err =
snd_pcm_hw_params_set_buffer_size_near(pcmPlaybackHandle,
hw_params,
&Audio_frames)) < 0) {
qDebug() << a1.
sprintf("Error setting buffer_size %d frames: %s",
Audio_frames, snd_strerror(err));
} else
qDebug() << a1.sprintf("setting Buffersize to %d --> %d: %s\n",
Audio_BufferSize, Audio_frames,
snd_strerror(err));
Audio_BufferSize = Audio_frames;
if ((err = snd_pcm_hw_params(pcmPlaybackHandle, hw_params)) < 0) {
qDebug() << a1.sprintf("Error setting HW params: %s",
snd_strerror(err));
}
Q_ASSERT(err >= 0);
}
void Audiooutput::ProduceAudioOutput(int n, int mmodes, int totalMModeGates,
short *sinusValue, short *cosinusValue)
{
for (int audioSample = 0; audioSample < n;
audioSample += Audio_ResamplingFactor) {
currentposition =
(int)(m_Audio.generalPos % (Audio_BufferSize / 2));
if (currentposition == 0) {
QueueAudioBuffer();
m_Audio.currentPos = 0;
}
m_Audio.generalPos++;
AudioData[currentposition * 2] =
(short)(sinusValue[audioSample]);
AudioData[currentposition * 2 + 1] =
(short)(cosinusValue[audioSample]);
}
}
void Audiooutput::QueueAudioBuffer()
{
snd_pcm_prepare(pcmPlaybackHandle);
Audio_DataWritten +=
snd_pcm_writei(pcmPlaybackHandle, AudioData, Audio_BufferSize);
}
Changing the audiobuffer size or fragments changes also the clicking period.
Can anyone help me with this issue ?
I checked also the first and Last Values. Thy are always difference.
OS: Ubuntu 11
more detail.
the count of received data is dynamically, and changes depend of different parameters. But I play always a certain part e.g. 128 values or 256 or 512....
// I get the Audiodata from a hardware (in a Timerloop)
audiobuffersize = 256;
short *AudioData = new short[256];
int generalAudioSample = 0;
void CollectDataFromHw()
{
...
int n = 0;
n = GetData(buf1,buf2);//buf1 = new short[MAX_SHRT]
if(n > 0)
FillAudioBuffer(n,buf1,buf2)
...
}
-------------------------------------------
void FillAudioBuffer(int n, short*buf1, short*buf2)
{
for(int audioSample = 0;audioSample < n; audioSample++){
iCurrentAudioSample = (int)(generalAudioSample % (audiobuffersize/2));
if(iCurrentAudioSample == 0) {
snd_pcm_writei(pcmPlaybackHandle,AudioData,audiobuffersize );
memset(AudioData,0x00,audiobuffersize*sizeof(short));
}
generalAudioSample++;
AudioData[iCurrentAudioSample * 2] = (short)(buf1[audioSample];
AudioData[iCurrentAudioSample * 2 +1] = (short)(buf2[audioSample];
}
}
I changed the audiobuffersize also. If I set it to a bigger size, I have some Echo additional to clicks.
any Idea ?
//-----------------------
the Problem is
snd_pcm_prepare(pcmPlaybackHandle);
every call of this function produce a click in sound !
Can't test the source code, but I think that the high-frequency clicks you hear are discontinuities in the sound wave. You have to assure that looping period (or, buffer size) is multiple of wave period.
Check if first and last value of buffer are almost the same (+/- 1, for example). Their distance determines the amplitude of the unwanted click.
solved
buffer has been played several times before it was filled with the data.
stupid error in the code.missing a parantez --> audio_buffersize/2 <--
and therefore the result was very often if(iCurrentAudioSample == 0) true !!!!!
iCurrentAudioSample = (int)(generalAudioSample % (audio_buffersize/2));
if(iCurrentAudioSample == 0)
{
writetoaudioStream(audiobuffer);
}