288 lines
9.3 KiB
C++
288 lines
9.3 KiB
C++
//
|
|
// libtgvoip is free and unencumbered public domain software.
|
|
// For more information, see http://unlicense.org or the UNLICENSE file
|
|
// you should have received with this source code distribution.
|
|
//
|
|
|
|
#include "OpusDecoder.h"
|
|
#include "audio/Resampler.h"
|
|
#include "logging.h"
|
|
#include <assert.h>
|
|
#include <math.h>
|
|
#include <algorithm>
|
|
#if defined HAVE_CONFIG_H || defined TGVOIP_USE_INSTALLED_OPUS
|
|
#include <opus/opus.h>
|
|
#else
|
|
#include "opus.h"
|
|
#endif
|
|
|
|
#include "VoIPController.h"
|
|
|
|
#define PACKET_SIZE (960 * 2)
|
|
|
|
using namespace tgvoip;
|
|
|
|
tgvoip::OpusDecoder::OpusDecoder(const std::shared_ptr<MediaStreamItf>& dst, bool isAsync, bool needEC) {
|
|
dst->SetCallback(OpusDecoder::Callback, this);
|
|
Initialize(isAsync, needEC);
|
|
}
|
|
|
|
tgvoip::OpusDecoder::OpusDecoder(const std::unique_ptr<MediaStreamItf>& dst, bool isAsync, bool needEC) {
|
|
dst->SetCallback(OpusDecoder::Callback, this);
|
|
Initialize(isAsync, needEC);
|
|
}
|
|
|
|
tgvoip::OpusDecoder::OpusDecoder(MediaStreamItf* dst, bool isAsync, bool needEC) {
|
|
dst->SetCallback(OpusDecoder::Callback, this);
|
|
Initialize(isAsync, needEC);
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::Initialize(bool isAsync, bool needEC) {
|
|
async = isAsync;
|
|
if (async) {
|
|
decodedQueue = new BlockingQueue<Buffer>(33);
|
|
semaphore = new Semaphore(32, 0);
|
|
} else {
|
|
decodedQueue = NULL;
|
|
semaphore = NULL;
|
|
}
|
|
dec=opus_decoder_create(48000, 1, NULL);
|
|
if (needEC)
|
|
ecDec = opus_decoder_create(48000, 1, NULL);
|
|
else
|
|
ecDec = NULL;
|
|
// todo buffer = reinterpret_cast<unsigned char*>(aligned_alloc(2, 8192));
|
|
buffer=(unsigned char *) malloc(8192);
|
|
lastDecoded = NULL;
|
|
outputBufferSize = 0;
|
|
echoCanceller = NULL;
|
|
frameDuration = 20;
|
|
consecutiveLostPackets = 0;
|
|
enableDTX = false;
|
|
silentPacketCount = 0;
|
|
levelMeter = NULL;
|
|
nextLen = 0;
|
|
running = false;
|
|
remainingDataLen = 0;
|
|
processedBuffer = NULL;
|
|
prevWasEC = false;
|
|
prevLastSample = 0;
|
|
}
|
|
|
|
tgvoip::OpusDecoder::~OpusDecoder() {
|
|
opus_decoder_destroy(dec);
|
|
if (ecDec)
|
|
opus_decoder_destroy(ecDec);
|
|
free(buffer);
|
|
if (decodedQueue)
|
|
delete decodedQueue;
|
|
if (semaphore)
|
|
delete semaphore;
|
|
}
|
|
|
|
|
|
void tgvoip::OpusDecoder::SetEchoCanceller(EchoCanceller* canceller) {
|
|
echoCanceller = canceller;
|
|
}
|
|
|
|
size_t tgvoip::OpusDecoder::Callback(unsigned char* data, size_t len, void* param) {
|
|
return (reinterpret_cast<OpusDecoder*>(param)->HandleCallback(data, len));
|
|
}
|
|
|
|
size_t tgvoip::OpusDecoder::HandleCallback(unsigned char* data, size_t len) {
|
|
if (async) {
|
|
if (!running) {
|
|
memset(data, 0, len);
|
|
return 0;
|
|
}
|
|
if (outputBufferSize == 0) {
|
|
outputBufferSize = len;
|
|
int packetsNeeded;
|
|
if (len > PACKET_SIZE)
|
|
packetsNeeded = len / PACKET_SIZE;
|
|
else
|
|
packetsNeeded = 1;
|
|
packetsNeeded *= 2;
|
|
semaphore->Release(packetsNeeded);
|
|
}
|
|
assert(outputBufferSize == len && "output buffer size is supposed to be the same throughout callbacks");
|
|
if (len == PACKET_SIZE) {
|
|
Buffer lastDecoded=decodedQueue->GetBlocking();
|
|
if (lastDecoded.IsEmpty())
|
|
return 0;
|
|
memcpy(data, *lastDecoded, PACKET_SIZE);
|
|
semaphore->Release();
|
|
if (silentPacketCount > 0) {
|
|
silentPacketCount--;
|
|
if (levelMeter)
|
|
levelMeter->Update(reinterpret_cast<int16_t*>(data), 0);
|
|
return 0;
|
|
}
|
|
if (echoCanceller) {
|
|
echoCanceller->SpeakerOutCallback(data, PACKET_SIZE);
|
|
}
|
|
} else {
|
|
LOGE("Opus decoder buffer length != 960 samples");
|
|
abort();
|
|
}
|
|
} else {
|
|
if (remainingDataLen == 0 && silentPacketCount == 0) {
|
|
int duration = DecodeNextFrame();
|
|
remainingDataLen = static_cast<size_t>(duration) / 20 * 960 * 2;
|
|
}
|
|
if (silentPacketCount > 0 || remainingDataLen == 0 || !processedBuffer){
|
|
if (silentPacketCount > 0)
|
|
silentPacketCount--;
|
|
memset(data, 0, 960 * 2);
|
|
if (levelMeter)
|
|
levelMeter->Update(reinterpret_cast<int16_t*>(data), 0);
|
|
return 0;
|
|
}
|
|
memcpy(data, processedBuffer, 960 * 2);
|
|
remainingDataLen -= 960 * 2;
|
|
if (remainingDataLen > 0) {
|
|
memmove(processedBuffer, processedBuffer + 960 * 2, remainingDataLen);
|
|
}
|
|
}
|
|
if (levelMeter)
|
|
levelMeter->Update(reinterpret_cast<int16_t*>(data), len / 2);
|
|
return len;
|
|
}
|
|
|
|
|
|
void tgvoip::OpusDecoder::Start() {
|
|
if (!async)
|
|
return;
|
|
running = true;
|
|
thread = new Thread(std::bind(&tgvoip::OpusDecoder::RunThread, this));
|
|
thread->SetName("opus_decoder");
|
|
thread->SetMaxPriority();
|
|
thread->Start();
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::Stop() {
|
|
if (!running || !async)
|
|
return;
|
|
running = false;
|
|
semaphore->Release();
|
|
thread->Join();
|
|
delete thread;
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::RunThread() {
|
|
LOGI("decoder: packets per frame %d", packetsPerFrame);
|
|
while(running) {
|
|
int playbackDuration = DecodeNextFrame();
|
|
for (int i = 0; i < playbackDuration / 20; i++) {
|
|
semaphore->Acquire();
|
|
if (!running) {
|
|
LOGI("==== decoder exiting ====");
|
|
return;
|
|
}
|
|
try {
|
|
Buffer buf=bufferPool.Get();
|
|
if (remainingDataLen > 0) {
|
|
for (effects::AudioEffect*& effect:postProcEffects) {
|
|
effect->Process(reinterpret_cast<int16_t*>(processedBuffer+(PACKET_SIZE * i)), 960);
|
|
}
|
|
buf.CopyFrom(processedBuffer + (PACKET_SIZE * i), 0, PACKET_SIZE);
|
|
} else {
|
|
//LOGE("Error decoding, result=%d", size);
|
|
memset(*buf, 0, PACKET_SIZE);
|
|
}
|
|
decodedQueue->Put(std::move(buf));
|
|
} catch (const std::bad_alloc&) {
|
|
LOGW("decoder: no buffers left!");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
int tgvoip::OpusDecoder::DecodeNextFrame() {
|
|
int playbackDuration = 0;
|
|
bool isEC = false;
|
|
size_t len = jitterBuffer->HandleOutput(buffer, 8192, 0, true, playbackDuration, isEC);
|
|
bool fec = false;
|
|
if (!len) {
|
|
fec = true;
|
|
len = jitterBuffer->HandleOutput(buffer, 8192, 0, false, playbackDuration, isEC);
|
|
//if(len)
|
|
// LOGV("Trying FEC...");
|
|
}
|
|
int size;
|
|
if (len) {
|
|
size = opus_decode(isEC ? ecDec : dec, buffer, len, reinterpret_cast<opus_int16*>(decodeBuffer), packetsPerFrame * 960, fec ? 1 : 0);
|
|
consecutiveLostPackets = 0;
|
|
if (prevWasEC != isEC && size) {
|
|
// It turns out the waveforms generated by the PLC feature are also great to help smooth out the
|
|
// otherwise audible transition between the frames from different decoders. Those are basically an extrapolation
|
|
// of the previous successfully decoded data -- which is exactly what we need here.
|
|
size = opus_decode(prevWasEC ? ecDec : dec, NULL, 0, reinterpret_cast<opus_int16*>(nextBuffer), packetsPerFrame * 960, 0);
|
|
if (size) {
|
|
int16_t* plcSamples = reinterpret_cast<int16_t*>(nextBuffer);
|
|
int16_t* samples = reinterpret_cast<int16_t*>(decodeBuffer);
|
|
constexpr float coeffs[] = {0.999802f, 0.995062f, 0.984031f, 0.966778f, 0.943413f, 0.914084f, 0.878975f, 0.838309f, 0.792344f, 0.741368f,
|
|
0.685706f, 0.625708f, 0.561754f, 0.494249f, 0.423619f, 0.350311f, 0.274788f, 0.197527f, 0.119018f, 0.039757f};
|
|
for (int i = 0; i < 20; i++) {
|
|
samples[i] = static_cast<int16_t>(round(plcSamples[i] * coeffs[i] + samples[i] * (1.f - coeffs[i])));
|
|
}
|
|
}
|
|
}
|
|
prevWasEC = isEC;
|
|
prevLastSample = decodeBuffer[size - 1];
|
|
} else { // do packet loss concealment
|
|
consecutiveLostPackets++;
|
|
if (consecutiveLostPackets > 2 && enableDTX) {
|
|
silentPacketCount += packetsPerFrame;
|
|
size = packetsPerFrame * 960;
|
|
} else {
|
|
size = opus_decode(prevWasEC ? ecDec : dec, NULL, 0, reinterpret_cast<opus_int16*>(decodeBuffer), packetsPerFrame * 960, 0);
|
|
//LOGV("PLC");
|
|
}
|
|
}
|
|
if (size < 0)
|
|
LOGW("decoder: opus_decode error %d", size);
|
|
remainingDataLen = size;
|
|
if (playbackDuration == 80) {
|
|
processedBuffer = buffer;
|
|
audio::Resampler::Rescale60To80(reinterpret_cast<int16_t*>(decodeBuffer),
|
|
reinterpret_cast<int16_t*>(processedBuffer));
|
|
} else if (playbackDuration == 40) {
|
|
processedBuffer = buffer;
|
|
audio::Resampler::Rescale60To40(reinterpret_cast<int16_t*>(decodeBuffer),
|
|
reinterpret_cast<int16_t*>(processedBuffer));
|
|
} else {
|
|
processedBuffer = decodeBuffer;
|
|
}
|
|
return playbackDuration;
|
|
}
|
|
|
|
|
|
void tgvoip::OpusDecoder::SetFrameDuration(uint32_t duration) {
|
|
frameDuration = duration;
|
|
packetsPerFrame = frameDuration / 20;
|
|
}
|
|
|
|
|
|
void tgvoip::OpusDecoder::SetJitterBuffer(std::shared_ptr<JitterBuffer> jitterBuffer) {
|
|
this->jitterBuffer = jitterBuffer;
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::SetDTX(bool enable) {
|
|
enableDTX = enable;
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::SetLevelMeter(AudioLevelMeter* levelMeter) {
|
|
this->levelMeter = levelMeter;
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::AddAudioEffect(effects::AudioEffect* effect) {
|
|
postProcEffects.push_back(effect);
|
|
}
|
|
|
|
void tgvoip::OpusDecoder::RemoveAudioEffect(effects::AudioEffect *effect) {
|
|
std::vector<effects::AudioEffect*>::iterator it = std::find(postProcEffects.begin(), postProcEffects.end(), effect);
|
|
if (it != postProcEffects.end())
|
|
postProcEffects.erase(it);
|
|
}
|