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Telegram: bump version

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This commit is contained in:
Gerasim Troeglazov
2025-05-26 19:12:08 +10:00
parent 48a726f434
commit cf774c6a5e
4 changed files with 1561 additions and 14784 deletions
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932
net-im/telegram-desktop/patches/libtgvoip-5.13.1.patchset
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@@ -1,932 +0,0 @@
From e1052847d80ec3c81d1166febe5c9149064e4e86 Mon Sep 17 00:00:00 2001
From: Gerasim Troeglazov <3dEyes@gmail.com>
Date: Thu, 3 Apr 2025 20:19:59 +1000
Subject: Add haiku support
diff --git a/Telegram/ThirdParty/libtgvoip/VoIPController.cpp b/Telegram/ThirdParty/libtgvoip/VoIPController.cpp
index 5a9731e..38e8bd7 100644
--- a/Telegram/ThirdParty/libtgvoip/VoIPController.cpp
+++ b/Telegram/ThirdParty/libtgvoip/VoIPController.cpp
@@ -8,6 +8,9 @@
#include <unistd.h>
#include <sys/time.h>
#endif
+#ifdef __HAIKU__
+#include <OS.h>
+#endif
#include <errno.h>
#include <string.h>
#include <wchar.h>
@@ -3015,6 +3018,10 @@ double VoIPController::GetCurrentTime(){
struct timespec ts;
clock_gettime(CLOCK_MONOTONIC, &ts);
return ts.tv_sec+(double)ts.tv_nsec/1000000000.0;
+#elif defined(__HAIKU__)
+ struct timeval tm;
+ gettimeofday(&tm, NULL);
+ return tm.tv_sec+(double)tm.tv_usec/1000000.0;
#elif defined(__APPLE__)
static pthread_once_t token = PTHREAD_ONCE_INIT;
pthread_once(&token, &initMachTimestart);
diff --git a/Telegram/ThirdParty/libtgvoip/audio/AudioIO.cpp b/Telegram/ThirdParty/libtgvoip/audio/AudioIO.cpp
index 8095646..ee4f1a8 100644
--- a/Telegram/ThirdParty/libtgvoip/audio/AudioIO.cpp
+++ b/Telegram/ThirdParty/libtgvoip/audio/AudioIO.cpp
@@ -39,6 +39,9 @@
#ifndef WITHOUT_PULSE
#include "../os/linux/AudioPulse.h"
#endif
+#elif defined(__HAIKU__)
+#include "../os/haiku/AudioInputHaiku.h"
+#include "../os/haiku/AudioOutputHaiku.h"
#else
#error "Unsupported operating system"
#endif
@@ -65,6 +68,8 @@ AudioIO* AudioIO::Create(std::string inputDevice, std::string outputDevice){
return new ContextlessAudioIO<AudioInputWave, AudioOutputWave>(inputDevice, outputDevice);
#endif
return new ContextlessAudioIO<AudioInputWASAPI, AudioOutputWASAPI>(inputDevice, outputDevice);
+#elif defined(__HAIKU__)
+ return new ContextlessAudioIO<AudioInputHaiku, AudioOutputHaiku>();
#elif defined(__linux__) || defined(__FreeBSD__) || defined(__OpenBSD__)
#ifndef WITHOUT_ALSA
#ifndef WITHOUT_PULSE
diff --git a/Telegram/ThirdParty/libtgvoip/audio/AudioInput.cpp b/Telegram/ThirdParty/libtgvoip/audio/AudioInput.cpp
index 1527497..90bbcf9 100644
--- a/Telegram/ThirdParty/libtgvoip/audio/AudioInput.cpp
+++ b/Telegram/ThirdParty/libtgvoip/audio/AudioInput.cpp
@@ -33,6 +33,8 @@
#ifndef WITHOUT_PULSE
#include "../os/linux/AudioPulse.h"
#endif
+#elif defined(__HAIKU__)
+#include "../os/haiku/AudioInputHaiku.h"
#else
#error "Unsupported operating system"
#endif
diff --git a/Telegram/ThirdParty/libtgvoip/audio/AudioOutput.cpp b/Telegram/ThirdParty/libtgvoip/audio/AudioOutput.cpp
index d42bd4d..3980367 100644
--- a/Telegram/ThirdParty/libtgvoip/audio/AudioOutput.cpp
+++ b/Telegram/ThirdParty/libtgvoip/audio/AudioOutput.cpp
@@ -37,6 +37,8 @@
#include "../os/linux/AudioOutputPulse.h"
#include "../os/linux/AudioPulse.h"
#endif
+#elif defined(__HAIKU__)
+#include "../os/haiku/AudioOutputHaiku.h"
#else
#error "Unsupported operating system"
#endif
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.cpp b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.cpp
new file mode 100644
index 0000000..7cce3e3
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.cpp
@@ -0,0 +1,276 @@
+//
+// 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 <stdlib.h>
+#include <stdio.h>
+#include <assert.h>
+#include <dlfcn.h>
+#include "AudioInputHaiku.h"
+#include "../../logging.h"
+#include "../../audio/Resampler.h"
+#include "../../VoIPController.h"
+
+#include "RingBuffer.h"
+
+using namespace tgvoip::audio;
+
+void RecordData(void* cookie, bigtime_t timestamp, void* data, size_t size, const media_format &format)
+{
+ AudioInputHaiku *audioInput = (AudioInputHaiku*)cookie;
+ if (!audioInput->IsRecording())
+ return;
+
+ if (format.u.raw_audio.format == media_raw_audio_format::B_AUDIO_SHORT &&
+ format.u.raw_audio.channel_count == 1) {
+ audioInput->fRingBuffer->Write((unsigned char*)data, size);
+ return;
+ }
+
+ uint32 bytesPerSample = 2;
+ switch (format.u.raw_audio.format) {
+ case media_raw_audio_format::B_AUDIO_CHAR:
+ bytesPerSample = 1;
+ break;
+ case media_raw_audio_format::B_AUDIO_SHORT:
+ bytesPerSample = 2;
+ break;
+ case media_raw_audio_format::B_AUDIO_INT:
+ bytesPerSample = 4;
+ break;
+ case media_raw_audio_format::B_AUDIO_FLOAT:
+ bytesPerSample = 4;
+ break;
+ default:
+ break;
+ }
+
+ int frames = size / (format.u.raw_audio.channel_count * bytesPerSample);
+ int16_t *dst = audioInput->workBuffer;
+
+ if (format.u.raw_audio.format == media_raw_audio_format::B_AUDIO_CHAR) {
+ unsigned char* src=reinterpret_cast<unsigned char*>(data);
+ for (int n=0; n < frames; n++) {
+ int32_t value = 0;
+ for (int j=0; j < format.u.raw_audio.channel_count; j++, src++) {
+ value += ((int32_t)(*src) - INT8_MAX) * UINT8_MAX;
+ }
+ value /= format.u.raw_audio.channel_count;
+ dst[n] = (int16_t)value;
+ }
+ } else if (format.u.raw_audio.format == media_raw_audio_format::B_AUDIO_SHORT) {
+ int16_t* src=reinterpret_cast<int16_t*>(data);
+ for (int n=0; n < frames; n++) {
+ int32_t value = 0;
+ for (int j=0; j < format.u.raw_audio.channel_count; j++, src++) {
+ value += *src;
+ }
+ value /= format.u.raw_audio.channel_count;
+ dst[n] = (int16_t)value;
+ }
+ } else if (format.u.raw_audio.format == media_raw_audio_format::B_AUDIO_INT) {
+ int32_t* src=reinterpret_cast<int32_t*>(data);
+ for (int n=0; n < frames; n++) {
+ int64_t value = 0;
+ for (int j=0; j < format.u.raw_audio.channel_count; j++, src++) {
+ value += (int64_t)(*src);
+ }
+ value /= format.u.raw_audio.channel_count;
+ dst[n] = (int16_t)(value / (UINT16_MAX + 1));
+ }
+ } else if (format.u.raw_audio.format == media_raw_audio_format::B_AUDIO_FLOAT) {
+ float* src=reinterpret_cast<float*>(data);
+ for (int n=0; n < frames; n++) {
+ float value = 0;
+ for (int j=0; j < format.u.raw_audio.channel_count; j++, src++) {
+ value += *src;
+ }
+ value /= format.u.raw_audio.channel_count;
+ dst[n] = (int16_t)(value*INT16_MAX);
+ }
+ }
+
+ if(format.u.raw_audio.frame_rate != audioInput->tgFrameRate) {
+ size_t len = tgvoip::audio::Resampler::Convert(dst, audioInput->convertBuffer,
+ frames, frames, audioInput->tgFrameRate, format.u.raw_audio.frame_rate) * audioInput->tgBytesPerSample;
+ audioInput->fRingBuffer->Write((unsigned char*)audioInput->convertBuffer, len);
+ } else {
+ audioInput->fRingBuffer->Write((unsigned char*)dst, frames * audioInput->tgBytesPerSample);
+ }
+}
+
+void NotifyRecordData(void * cookie, BMediaRecorder::notification code, ...)
+{
+ AudioInputHaiku *audioInput = (AudioInputHaiku*)cookie;
+ if (code == BMediaRecorder::B_WILL_STOP) {
+ if (audioInput->IsRecording()) {
+ audioInput->Stop();
+ }
+ }
+}
+
+AudioInputHaiku::AudioInputHaiku()
+{
+ fRecorder = NULL;
+ fRingBuffer = NULL;
+ isRecording = false;
+
+ tgFrameRate = 48000;
+ tgChannelsCount = 1;
+ tgBytesPerSample = 2;
+
+ status_t error;
+
+ fRoster = BMediaRoster::Roster(&error);
+ if (!fRoster) {
+ failed=true;
+ return;
+ }
+ error = fRoster->GetAudioInput(&fAudioInputNode);
+ if (error < B_OK) {
+ failed=true;
+ return;
+ }
+ error = fRoster->GetAudioMixer(&fAudioMixerNode);
+ if (error < B_OK) {
+ failed=true;
+ return;
+ }
+ fRecorder = new BMediaRecorder("Telegram", B_MEDIA_RAW_AUDIO);
+ if (fRecorder->InitCheck() < B_OK) {
+ failed=true;
+ return;
+ }
+ media_format output_format;
+ output_format.type = B_MEDIA_RAW_AUDIO;
+ output_format.u.raw_audio = media_raw_audio_format::wildcard;
+ output_format.u.raw_audio.channel_count = 1;
+ fRecorder->SetAcceptedFormat(output_format);
+
+ const int maxInputCount = 64;
+ dormant_node_info dni[maxInputCount];
+
+ int32 real_count = maxInputCount;
+
+ error = fRoster->GetDormantNodes(dni, &real_count, 0, &output_format, 0, B_BUFFER_PRODUCER | B_PHYSICAL_INPUT);
+ if (real_count > maxInputCount)
+ real_count = maxInputCount;
+ char selected_name[B_MEDIA_NAME_LENGTH] = "Default input";
+
+ for (int i = 0; i < real_count; i++) {
+ media_node_id ni[12];
+ int32 ni_count = 12;
+ error = fRoster->GetInstancesFor(dni[i].addon, dni[i].flavor_id, ni, &ni_count);
+ if (error == B_OK) {
+ for (int j = 0; j < ni_count; j++) {
+ if (ni[j] == fAudioInputNode.node) {
+ strcpy(selected_name, dni[i].name);
+ break;
+ }
+ }
+ }
+ }
+
+ media_output audioOutput;
+ if (!fRecorder->IsConnected()) {
+ int32 count = 0;
+ error = fRoster->GetFreeOutputsFor(fAudioInputNode, &audioOutput, 1, &count, B_MEDIA_RAW_AUDIO);
+ if (error < B_OK) {
+ failed=true;
+ return;
+ }
+
+ if (count < 1) {
+ failed=true;
+ return;
+ }
+ fRecordFormat.u.raw_audio = audioOutput.format.u.raw_audio;
+ } else {
+ fRecordFormat.u.raw_audio = fRecorder->AcceptedFormat().u.raw_audio;
+ }
+ fRecordFormat.type = B_MEDIA_RAW_AUDIO;
+
+ error = fRecorder->SetHooks(RecordData, NotifyRecordData, this);
+ if (error < B_OK) {
+ failed=true;
+ return;
+ }
+
+ if (!fRecorder->IsConnected()) {
+ error = fRecorder->Connect(fAudioInputNode, &audioOutput, &fRecordFormat);
+ if (error < B_OK) {
+ fRecorder->SetHooks(NULL, NULL, NULL);
+ failed=true;
+ return;
+ }
+ }
+
+ fRingBuffer = new RingBuffer(BUFFER_SIZE * 2 * 3);
+ if (fRingBuffer->InitCheck() != B_OK) {
+ failed=true;
+ return;
+ }
+}
+
+AudioInputHaiku::~AudioInputHaiku(){
+ if (fRecorder != NULL) {
+ if (fRecorder->InitCheck() == B_OK) {
+ if (fRecorder->IsConnected())
+ fRecorder->Disconnect();
+ }
+ delete fRecorder;
+ }
+ if (fRingBuffer != NULL)
+ delete fRingBuffer;
+}
+
+void AudioInputHaiku::Configure(uint32_t sampleRate, uint32_t bitsPerSample, uint32_t channels){
+ tgFrameRate = sampleRate;
+ tgChannelsCount = channels;
+ tgBytesPerSample = bitsPerSample / 8;
+}
+
+bool AudioInputHaiku::IsRecording(){
+ return isRecording;
+}
+
+void AudioInputHaiku::Start(){
+ if(failed || isRecording)
+ return;
+
+ isRecording=true;
+
+ thread = new Thread(std::bind(&AudioInputHaiku::RunThread, this));
+ thread->SetName("AudioInputHaiku");
+ thread->Start();
+
+ fRecorder->Start();
+}
+
+void AudioInputHaiku::Stop(){
+ if(!isRecording)
+ return;
+
+ isRecording=false;
+
+ fRecorder->Stop();
+
+ thread->Join();
+ delete thread;
+ thread=NULL;
+}
+
+void AudioInputHaiku::RunThread(){
+ unsigned char buffer[BUFFER_SIZE*2];
+ while (isRecording){
+ if (fRingBuffer->GetReadAvailable() >= sizeof(buffer)) {
+ int readed = fRingBuffer->Read(buffer, sizeof(buffer));
+ if (readed < sizeof(buffer))
+ memset(buffer + readed, 0, sizeof(buffer) - readed);
+ InvokeCallback(buffer, sizeof(buffer));
+ } else
+ snooze(100);
+ }
+}
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.h b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.h
new file mode 100644
index 0000000..1c63afe
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioInputHaiku.h
@@ -0,0 +1,66 @@
+//
+// 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.
+//
+
+#ifndef LIBTGVOIP_AUDIOINPUTHAIKU_H
+#define LIBTGVOIP_AUDIOINPUTHAIKU_H
+
+#include "../../audio/AudioInput.h"
+#include "../../threading.h"
+
+#include <OS.h>
+#include <MediaFile.h>
+#include <MediaNode.h>
+#include <MediaRecorder.h>
+#include <MediaTrack.h>
+#include <MediaRoster.h>
+#include <TimeSource.h>
+#include <NodeInfo.h>
+#include <MediaAddOn.h>
+
+#include "RingBuffer.h"
+
+#define BUFFER_SIZE 960
+
+namespace tgvoip{
+namespace audio{
+
+class AudioInputHaiku : public AudioInput{
+
+public:
+ AudioInputHaiku();
+ virtual ~AudioInputHaiku();
+ virtual void Configure(uint32_t sampleRate, uint32_t bitsPerSample, uint32_t channels);
+ virtual void Start();
+ virtual void Stop();
+ virtual bool IsRecording();
+
+ RingBuffer *fRingBuffer;
+ int16_t workBuffer[BUFFER_SIZE * 64];
+ int16_t convertBuffer[BUFFER_SIZE * 64];
+
+ uint32 tgFrameRate;
+ uint32 tgChannelsCount;
+ uint32 tgBytesPerSample;
+
+private:
+ void RunThread();
+
+ bool isConfigured;
+ bool isRecording;
+
+ BMediaRoster * fRoster;
+ BMediaRecorder * fRecorder;
+ media_format fRecordFormat;
+ media_node fAudioInputNode;
+ media_node fAudioMixerNode;
+
+ Thread* thread;
+};
+
+}
+}
+
+#endif //LIBTGVOIP_AUDIOINPUTHAIKU_H
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.cpp b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.cpp
new file mode 100644
index 0000000..2fca8a1
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.cpp
@@ -0,0 +1,99 @@
+//
+// 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 <assert.h>
+#include <dlfcn.h>
+#include "AudioOutputHaiku.h"
+#include "../../logging.h"
+#include "../../VoIPController.h"
+
+#define BUFFER_SIZE 960
+
+using namespace tgvoip::audio;
+
+static void playerProc(void *cookie, void *buffer, size_t len, const media_raw_audio_format &format)
+{
+ AudioOutputHaiku *obj = (AudioOutputHaiku*)cookie;
+ obj->InvokeCallback((unsigned char*)buffer, len);
+}
+
+
+AudioOutputHaiku::AudioOutputHaiku(){
+ soundPlayer = NULL;
+ isPlaying = false;
+ isConfigured = false;
+ Configure(48000, 16, 1);
+ return;
+}
+
+AudioOutputHaiku::~AudioOutputHaiku(){
+ if (isConfigured) {
+ if (soundPlayer != NULL) {
+ soundPlayer->Stop();
+ delete soundPlayer;
+ }
+ }
+}
+
+void AudioOutputHaiku::Configure(uint32_t sampleRate, uint32_t bitsPerSample, uint32_t channels){
+ media_raw_audio_format mediaKitFormat = {
+ (float)sampleRate,
+ (uint32)channels,
+ media_raw_audio_format::B_AUDIO_SHORT,
+ B_MEDIA_LITTLE_ENDIAN,
+ (uint32)BUFFER_SIZE * (bitsPerSample / 8) * channels
+ };
+
+ switch (bitsPerSample) {
+ case 8:
+ mediaKitFormat.format = media_raw_audio_format::B_AUDIO_CHAR;
+ break;
+ case 16:
+ mediaKitFormat.format = media_raw_audio_format::B_AUDIO_SHORT;
+ break;
+ case 32:
+ mediaKitFormat.format = media_raw_audio_format::B_AUDIO_INT;
+ break;
+ default:
+ mediaKitFormat.format = media_raw_audio_format::B_AUDIO_SHORT;
+ break;
+ }
+
+ soundPlayer = new BSoundPlayer(&mediaKitFormat, "Telegram", playerProc, NULL, (void*)this);
+
+ if(soundPlayer->InitCheck() != B_OK) {
+ delete soundPlayer;
+ soundPlayer = NULL;
+ isPlaying = false;
+ failed = true;
+ return;
+ }
+
+ isConfigured = true;
+}
+
+void AudioOutputHaiku::Start(){
+ if(soundPlayer == NULL || isPlaying)
+ return;
+
+ soundPlayer->Start();
+ soundPlayer->SetHasData(true);
+
+ isPlaying=true;
+}
+
+void AudioOutputHaiku::Stop(){
+ if(!isPlaying)
+ return;
+
+ soundPlayer->Stop();
+ isPlaying=false;
+}
+
+bool AudioOutputHaiku::IsPlaying(){
+ return isPlaying;
+}
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.h b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.h
new file mode 100644
index 0000000..91f2521
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/AudioOutputHaiku.h
@@ -0,0 +1,35 @@
+//
+// 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.
+//
+
+#ifndef LIBTGVOIP_AUDIOOUTPUTHAIKU_H
+#define LIBTGVOIP_AUDIOOUTPUTHAIKU_H
+
+#include "../../audio/AudioOutput.h"
+#include "../../threading.h"
+
+#include <SoundPlayer.h>
+
+namespace tgvoip{
+namespace audio{
+
+class AudioOutputHaiku : public AudioOutput{
+public:
+ AudioOutputHaiku();
+ virtual ~AudioOutputHaiku();
+ virtual void Configure(uint32_t sampleRate, uint32_t bitsPerSample, uint32_t channels);
+ virtual void Start();
+ virtual void Stop();
+ virtual bool IsPlaying() override;
+private:
+ bool isPlaying;
+ bool isConfigured;
+ BSoundPlayer *soundPlayer;
+};
+
+}
+}
+
+#endif //LIBTGVOIP_AUDIOOUTPUTHAIKU_H
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.cpp b/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.cpp
new file mode 100644
index 0000000..6c94933
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.cpp
@@ -0,0 +1,136 @@
+//
+// 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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <OS.h>
+
+#include "RingBuffer.h"
+
+RingBuffer::RingBuffer( int size )
+{
+ initialized = false;
+ Buffer = new unsigned char[size];
+ if(Buffer!=NULL) {
+ memset( Buffer, 0, size );
+ BufferSize = size;
+ } else {
+ BufferSize = 0;
+ }
+ reader = 0;
+ writer = 0;
+ writeBytesAvailable = size;
+ if((locker=create_sem(1,"locker")) >= B_OK) {
+ initialized = true;
+ } else {
+ if(Buffer!=NULL) {
+ delete[] Buffer;
+ }
+ }
+}
+
+RingBuffer::~RingBuffer( )
+{
+ if(initialized) {
+ delete[] Buffer;
+ delete_sem(locker);
+ }
+}
+
+bool
+RingBuffer::Empty( void )
+{
+ memset( Buffer, 0, BufferSize );
+ reader = 0;
+ writer = 0;
+ writeBytesAvailable = BufferSize;
+ return true;
+}
+
+int
+RingBuffer::Read( unsigned char *data, int size )
+{
+ acquire_sem(locker);
+
+ if( data == 0 || size <= 0 || writeBytesAvailable == BufferSize ) {
+ release_sem(locker);
+ return 0;
+ }
+
+ int readBytesAvailable = BufferSize - writeBytesAvailable;
+
+ if( size > readBytesAvailable ) {
+ size = readBytesAvailable;
+ }
+
+ if(size > BufferSize - reader) {
+ int len = BufferSize - reader;
+ memcpy(data, Buffer + reader, len);
+ memcpy(data + len, Buffer, size-len);
+ } else {
+ memcpy(data, Buffer + reader, size);
+ }
+
+ reader = (reader + size) % BufferSize;
+ writeBytesAvailable += size;
+
+ release_sem(locker);
+ return size;
+}
+
+int
+RingBuffer::Write( unsigned char *data, int size )
+{
+ acquire_sem(locker);
+
+ if( data == 0 || size <= 0 || writeBytesAvailable == 0 ) {
+ release_sem(locker);
+ return 0;
+ }
+
+ if( size > writeBytesAvailable ) {
+ size = writeBytesAvailable;
+ }
+
+ if(size > BufferSize - writer) {
+ int len = BufferSize - writer;
+ memcpy(Buffer + writer, data, len);
+ memcpy(Buffer, data+len, size-len);
+ } else {
+ memcpy(Buffer + writer, data, size);
+ }
+
+ writer = (writer + size) % BufferSize;
+ writeBytesAvailable -= size;
+
+ release_sem(locker);
+ return size;
+}
+
+int
+RingBuffer::GetSize( void )
+{
+ return BufferSize;
+}
+
+int
+RingBuffer::GetWriteAvailable( void )
+{
+ return writeBytesAvailable;
+}
+
+int
+RingBuffer::GetReadAvailable( void )
+{
+ return BufferSize - writeBytesAvailable;
+}
+
+status_t
+RingBuffer::InitCheck( void )
+{
+ return initialized?B_OK:B_ERROR;
+}
diff --git a/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.h b/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.h
new file mode 100644
index 0000000..01f6096
--- /dev/null
+++ b/Telegram/ThirdParty/libtgvoip/os/haiku/RingBuffer.h
@@ -0,0 +1,37 @@
+//
+// 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.
+//
+
+#ifndef __RING_BUFFER_H__
+#define __RING_BUFFER_H__
+
+#include <OS.h>
+
+class RingBuffer {
+
+public:
+ RingBuffer(int size);
+ ~RingBuffer();
+ int Read( unsigned char* dataPtr, int numBytes );
+ int Write( unsigned char *dataPtr, int numBytes );
+
+ bool Empty( void );
+ int GetSize( );
+ int GetWriteAvailable( );
+ int GetReadAvailable( );
+ status_t InitCheck( );
+private:
+ unsigned char *Buffer;
+ int BufferSize;
+ int reader;
+ int writer;
+ int writeBytesAvailable;
+
+ sem_id locker;
+
+ bool initialized;
+};
+
+#endif
diff --git a/Telegram/ThirdParty/libtgvoip/os/posix/NetworkSocketPosix.cpp b/Telegram/ThirdParty/libtgvoip/os/posix/NetworkSocketPosix.cpp
index 78e0583..81bf9fc 100644
--- a/Telegram/ThirdParty/libtgvoip/os/posix/NetworkSocketPosix.cpp
+++ b/Telegram/ThirdParty/libtgvoip/os/posix/NetworkSocketPosix.cpp
@@ -248,12 +248,13 @@ void NetworkSocketPosix::Open(){
}
int flag=0;
int res=setsockopt(fd, IPPROTO_IPV6, IPV6_V6ONLY, &flag, sizeof(flag));
+#ifndef __HAIKU__
if(res<0){
LOGE("error enabling dual stack socket: %d / %s", errno, strerror(errno));
failed=true;
return;
}
-
+#endif
SetMaxPriority();
fcntl(fd, F_SETFL, O_NONBLOCK);
@@ -403,6 +404,8 @@ std::string NetworkSocketPosix::GetLocalInterfaceInfo(IPv4Address *v4addr, IPv6A
if(didAttach){
sharedJVM->DetachCurrentThread();
}
+#elif defined(__HAIKU__)
+ return name;
#else
struct ifaddrs* interfaces;
if(!getifaddrs(&interfaces)){
diff --git a/Telegram/ThirdParty/libtgvoip/threading.h b/Telegram/ThirdParty/libtgvoip/threading.h
old mode 100755
new mode 100644
index 7da4fb2..a96bc52
--- a/Telegram/ThirdParty/libtgvoip/threading.h
+++ b/Telegram/ThirdParty/libtgvoip/threading.h
@@ -9,7 +9,7 @@
#include <functional>
-#if defined(_POSIX_THREADS) || defined(_POSIX_VERSION) || defined(__unix__) || defined(__unix) || (defined(__APPLE__) && defined(__MACH__))
+#if defined(_POSIX_THREADS) || defined(_POSIX_VERSION) || defined(__unix__) || defined(__unix) || defined(__HAIKU__) || (defined(__APPLE__) && defined(__MACH__))
#include <pthread.h>
#include <semaphore.h>
@@ -94,6 +94,7 @@ namespace tgvoip{
static void* ActualEntryPoint(void* arg){
Thread* self=reinterpret_cast<Thread*>(arg);
if(self->name){
+#ifndef __HAIKU__
#if defined(__linux__) || defined(__FreeBSD__)
pthread_setname_np(self->thread, self->name);
#elif defined(__OpenBSD__)
@@ -104,6 +105,7 @@ namespace tgvoip{
DarwinSpecific::SetCurrentThreadPriority(DarwinSpecific::THREAD_PRIO_USER_INTERACTIVE);
}
#endif
+#endif //__HAIKU__
}
self->entry();
return NULL;
diff --git a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/logging_webrtc.cc b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/logging_webrtc.cc
old mode 100755
new mode 100644
index a8d1522..991241b
--- a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/logging_webrtc.cc
+++ b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/logging_webrtc.cc
@@ -28,6 +28,10 @@
static const int kMaxLogLineSize = 1024 - 60;
#endif // WEBRTC_MAC && !defined(WEBRTC_IOS) || WEBRTC_ANDROID
+#if defined(WEBRTC_HAIKU)
+#include <OS.h>
+#endif
+
#include <stdio.h>
#include <string.h>
#include <time.h>
@@ -120,7 +124,12 @@ LogMessage::LogMessage(const char* file,
if (thread_) {
PlatformThreadId id = CurrentThreadId();
+#if defined(WEBRTC_HAIKU)
+ thread_id tid = get_pthread_thread_id(id);
+ print_stream_ << "[" << tid << "] ";
+#else
print_stream_ << "[" << id << "] ";
+#endif
}
if (file != nullptr) {
diff --git a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_file.h b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_file.h
old mode 100755
new mode 100644
diff --git a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.cc b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.cc
index cf7d478..f27b9a1 100644
--- a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.cc
+++ b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.cc
@@ -20,6 +20,8 @@ namespace rtc {
PlatformThreadId CurrentThreadId() {
#if defined(WEBRTC_WIN)
return GetCurrentThreadId();
+#elif defined(WEBRTC_HAIKU)
+ return pthread_self();
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
return pthread_mach_thread_np(pthread_self());
diff --git a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.h b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.h
index 0bc42eb..c87cde9 100644
--- a/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.h
+++ b/Telegram/ThirdParty/libtgvoip/webrtc_dsp/rtc_base/platform_thread_types.h
@@ -35,6 +35,9 @@ typedef DWORD PlatformThreadRef;
#elif defined(WEBRTC_FUCHSIA)
typedef zx_handle_t PlatformThreadId;
typedef zx_handle_t PlatformThreadRef;
+#elif defined(WEBRTC_HAIKU)
+typedef pthread_t PlatformThreadId;
+typedef pthread_t PlatformThreadRef;
#elif defined(WEBRTC_POSIX)
typedef pid_t PlatformThreadId;
typedef pthread_t PlatformThreadRef;
diff --git a/Telegram/cmake/lib_tgvoip.cmake b/Telegram/cmake/lib_tgvoip.cmake
index fbae709..18f96ec 100644
--- a/Telegram/cmake/lib_tgvoip.cmake
+++ b/Telegram/cmake/lib_tgvoip.cmake
@@ -118,6 +118,14 @@ PRIVATE
os/linux/AudioPulse.cpp
os/linux/AudioPulse.h
+ # Haiku
+ os/haiku/AudioInputHaiku.cpp
+ os/haiku/AudioInputHaiku.h
+ os/haiku/AudioOutputHaiku.cpp
+ os/haiku/AudioOutputHaiku.h
+ os/haiku/RingBuffer.cpp
+ os/haiku/RingBuffer.h
+
# POSIX
os/posix/NetworkSocketPosix.cpp
os/posix/NetworkSocketPosix.h
@@ -153,6 +161,25 @@ elseif (APPLE)
TGVOIP_NO_OSX_PRIVATE_API
)
endif()
+elseif (HAIKU)
+ target_compile_definitions(lib_tgvoip_bundled
+ PUBLIC
+ WEBRTC_POSIX
+ WEBRTC_HAIKU
+ )
+ target_compile_options(lib_tgvoip_bundled
+ PRIVATE
+ -Wno-unknown-pragmas
+ -Wno-error=sequence-point
+ -Wno-error=unused-result
+ -mmmx
+ -msse2
+ )
+ target_link_libraries(lib_tgvoip_bundled
+ PRIVATE
+ network
+ media
+ )
else()
add_library(lib_tgvoip_bundled_options INTERFACE)
target_compile_options(lib_tgvoip_bundled_options
--
2.48.1

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net-im/telegram-desktop/patches/pch_disable-5.13.1.patchset → net-im/telegram-desktop/patches/pch_disable-5.14.3.patchset
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