mirror of
https://github.com/raysan5/raylib.git
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Removed useless spaces
This commit is contained in:
raysan5
288
src/audio.c
288
src/audio.c
@ -3,21 +3,21 @@
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* raylib.audio
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*
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* Basic functions to manage Audio: InitAudioDevice, LoadAudioFiles, PlayAudioFiles
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*
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* Uses external lib:
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*
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* Uses external lib:
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* OpenAL - Audio device management lib
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* stb_vorbis - Ogg audio files loading
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*
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*
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* Copyright (c) 2013 Ramon Santamaria (Ray San - raysan@raysanweb.com)
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*
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* This software is provided "as-is", without any express or implied warranty. In no event
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*
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* This software is provided "as-is", without any express or implied warranty. In no event
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* will the authors be held liable for any damages arising from the use of this software.
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*
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* Permission is granted to anyone to use this software for any purpose, including commercial
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* Permission is granted to anyone to use this software for any purpose, including commercial
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* applications, and to alter it and redistribute it freely, subject to the following restrictions:
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*
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* 1. The origin of this software must not be misrepresented; you must not claim that you
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* wrote the original software. If you use this software in a product, an acknowledgment
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* 1. The origin of this software must not be misrepresented; you must not claim that you
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* wrote the original software. If you use this software in a product, an acknowledgment
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* in the product documentation would be appreciated but is not required.
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*
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* 2. Altered source versions must be plainly marked as such, and must not be misrepresented
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@ -54,16 +54,16 @@
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// NOTE: Anything longer than ~10 seconds should be streamed...
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typedef struct Music {
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stb_vorbis *stream;
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ALuint buffers[MUSIC_STREAM_BUFFERS];
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ALuint source;
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ALenum format;
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ALuint buffers[MUSIC_STREAM_BUFFERS];
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ALuint source;
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ALenum format;
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int channels;
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int sampleRate;
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int totalSamplesLeft;
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bool loop;
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int totalSamplesLeft;
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bool loop;
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} Music;
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// Wave file data
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@ -72,7 +72,7 @@ typedef struct Wave {
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unsigned int dataSize; // Data size in bytes
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unsigned int sampleRate;
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short bitsPerSample;
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short channels;
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short channels;
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} Wave;
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//----------------------------------------------------------------------------------
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@ -102,22 +102,22 @@ void InitAudioDevice()
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{
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// Open and initialize a device with default settings
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ALCdevice *device = alcOpenDevice(NULL);
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if(!device) TraceLog(ERROR, "Could not open audio device");
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ALCcontext *context = alcCreateContext(device, NULL);
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if(context == NULL || alcMakeContextCurrent(context) == ALC_FALSE)
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{
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if(context != NULL) alcDestroyContext(context);
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alcCloseDevice(device);
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TraceLog(ERROR, "Could not setup audio context");
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}
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TraceLog(INFO, "Audio device and context initialized successfully: %s\n", alcGetString(device, ALC_DEVICE_SPECIFIER));
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// Listener definition (just for 2D)
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alListener3f(AL_POSITION, 0, 0, 0);
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alListener3f(AL_VELOCITY, 0, 0, 0);
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@ -131,7 +131,7 @@ void CloseAudioDevice()
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ALCdevice *device;
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ALCcontext *context = alcGetCurrentContext();
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if (context == NULL) TraceLog(WARNING, "Could not get current audio context for closing");
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device = alcGetContextsDevice(context);
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@ -150,41 +150,41 @@ Sound LoadSound(char *fileName)
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{
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Sound sound;
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Wave wave;
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// NOTE: The entire file is loaded to memory to play it all at once (no-streaming)
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// Audio file loading
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// NOTE: Buffer space is allocated inside function, Wave must be freed
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if (strcmp(GetExtension(fileName),"wav") == 0) wave = LoadWAV(fileName);
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else if (strcmp(GetExtension(fileName),"ogg") == 0) wave = LoadOGG(fileName);
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else TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
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if (wave.data != NULL)
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{
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ALenum format = 0;
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// The OpenAL format is worked out by looking at the number of channels and the bits per sample
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if (wave.channels == 1)
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if (wave.channels == 1)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
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}
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else if (wave.channels == 2)
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}
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else if (wave.channels == 2)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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ALuint source;
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alGenSources(1, &source); // Generate pointer to audio source
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_GAIN, 1);
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alSource3f(source, AL_POSITION, 0, 0, 0);
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alSource3f(source, AL_VELOCITY, 0, 0, 0);
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alSourcei(source, AL_LOOPING, AL_FALSE);
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// Convert loaded data to OpenAL buffer
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//----------------------------------------
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ALuint buffer;
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@ -195,17 +195,17 @@ Sound LoadSound(char *fileName)
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// Attach sound buffer to source
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alSourcei(source, AL_BUFFER, buffer);
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// Unallocate WAV data
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UnloadWave(wave);
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TraceLog(INFO, "[%s] Sound file loaded successfully", fileName);
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TraceLog(INFO, "[%s] Sound file loaded successfully", fileName);
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TraceLog(INFO, "[%s] Sample rate: %i - Channels: %i", fileName, wave.sampleRate, wave.channels);
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sound.source = source;
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sound.buffer = buffer;
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}
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return sound;
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}
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@ -220,9 +220,9 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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unsigned char version; // rRES file version and subversion
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char useless; // rRES header reserved data
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short numRes;
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ResInfoHeader infoHeader;
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FILE *rresFile = fopen(rresName, "rb");
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if (!rresFile) TraceLog(WARNING, "[%s] Could not open raylib resource file", rresName);
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@ -235,7 +235,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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fread(&id[3], sizeof(char), 1, rresFile);
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fread(&version, sizeof(char), 1, rresFile);
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fread(&useless, sizeof(char), 1, rresFile);
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if ((id[0] != 'r') && (id[1] != 'R') && (id[2] != 'E') &&(id[3] != 'S'))
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{
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TraceLog(WARNING, "[%s] This is not a valid raylib resource file", rresName);
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@ -244,11 +244,11 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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{
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// Read number of resources embedded
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fread(&numRes, sizeof(short), 1, rresFile);
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for (int i = 0; i < numRes; i++)
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{
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fread(&infoHeader, sizeof(ResInfoHeader), 1, rresFile);
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if (infoHeader.id == resId)
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{
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found = true;
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@ -258,56 +258,56 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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{
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// TODO: Check data compression type
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// NOTE: We suppose compression type 2 (DEFLATE - default)
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// Reading SOUND parameters
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Wave wave;
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short sampleRate, bps;
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char channels, reserved;
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fread(&sampleRate, sizeof(short), 1, rresFile); // Sample rate (frequency)
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fread(&bps, sizeof(short), 1, rresFile); // Bits per sample
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fread(&channels, 1, 1, rresFile); // Channels (1 - mono, 2 - stereo)
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fread(&reserved, 1, 1, rresFile); // <reserved>
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wave.sampleRate = sampleRate;
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wave.dataSize = infoHeader.srcSize;
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wave.bitsPerSample = bps;
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wave.channels = (short)channels;
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unsigned char *data = malloc(infoHeader.size);
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fread(data, infoHeader.size, 1, rresFile);
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wave.data = DecompressData(data, infoHeader.size, infoHeader.srcSize);
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free(data);
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// Convert wave to Sound (OpenAL)
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ALenum format = 0;
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// The OpenAL format is worked out by looking at the number of channels and the bits per sample
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if (wave.channels == 1)
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if (wave.channels == 1)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_MONO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_MONO16;
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}
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else if (wave.channels == 2)
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}
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else if (wave.channels == 2)
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{
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if (wave.bitsPerSample == 8 ) format = AL_FORMAT_STEREO8;
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else if (wave.bitsPerSample == 16) format = AL_FORMAT_STEREO16;
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}
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// Create an audio source
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ALuint source;
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alGenSources(1, &source); // Generate pointer to audio source
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_PITCH, 1);
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alSourcef(source, AL_GAIN, 1);
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alSource3f(source, AL_POSITION, 0, 0, 0);
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alSource3f(source, AL_VELOCITY, 0, 0, 0);
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alSourcei(source, AL_LOOPING, AL_FALSE);
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// Convert loaded data to OpenAL buffer
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//----------------------------------------
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ALuint buffer;
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@ -318,12 +318,12 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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// Attach sound buffer to source
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alSourcei(source, AL_BUFFER, buffer);
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// Unallocate WAV data
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UnloadWave(wave);
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TraceLog(INFO, "[%s] Sound loaded successfully from resource, sample rate: %i", rresName, (int)sampleRate);
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sound.source = source;
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sound.buffer = buffer;
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}
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@ -344,18 +344,18 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
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case 4: break; // RAW: No parameters
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default: break;
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}
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// Jump DATA to read next infoHeader
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fseek(rresFile, infoHeader.size, SEEK_CUR);
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}
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}
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}
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}
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fclose(rresFile);
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}
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if (!found) TraceLog(WARNING, "[%s] Required resource id [%i] could not be found in the raylib resource file", rresName, resId);
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return sound;
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}
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@ -370,7 +370,7 @@ void UnloadSound(Sound sound)
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void PlaySound(Sound sound)
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{
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alSourcePlay(sound.source); // Play the sound
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//TraceLog(INFO, "Playing sound");
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// Find the current position of the sound being played
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@ -380,7 +380,7 @@ void PlaySound(Sound sound)
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//
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//int sampleRate;
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//alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); // AL_CHANNELS, AL_BITS (bps)
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//float seconds = (float)byteOffset / sampleRate; // Number of seconds since the beginning of the sound
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//or
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//float result;
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@ -404,10 +404,10 @@ bool SoundIsPlaying(Sound sound)
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{
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bool playing = false;
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ALint state;
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alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
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if (state == AL_PLAYING) playing = true;
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return playing;
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}
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@ -434,49 +434,49 @@ void PlayMusicStream(char *fileName)
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{
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// Stop current music, clean buffers, unload current stream
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StopMusicStream();
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// Open audio stream
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currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL);
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if (currentMusic.stream == NULL) TraceLog(WARNING, "[%s] Could not open ogg audio file", fileName);
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else
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{
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// Get file info
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stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream);
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currentMusic.channels = info.channels;
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currentMusic.sampleRate = info.sample_rate;
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TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
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TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels);
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TraceLog(INFO, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
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if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16;
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else currentMusic.format = AL_FORMAT_MONO16;
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currentMusic.loop = true; // We loop by default
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musicEnabled = true;
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// Create an audio source
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alGenSources(1, ¤tMusic.source); // Generate pointer to audio source
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alSourcef(currentMusic.source, AL_PITCH, 1);
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alSourcef(currentMusic.source, AL_PITCH, 1);
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alSourcef(currentMusic.source, AL_GAIN, 1);
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alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0);
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alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0);
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//alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue!
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// Generate two OpenAL buffers
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alGenBuffers(2, currentMusic.buffers);
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// Fill buffers with music...
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BufferMusicStream(currentMusic.buffers[0]);
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BufferMusicStream(currentMusic.buffers[1]);
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// Queue buffers and start playing
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alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers);
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alSourcePlay(currentMusic.source);
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// NOTE: Regularly, we must check if a buffer has been processed and refill it: MusicStreamUpdate()
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currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
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@ -491,15 +491,15 @@ void StopMusicStream()
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if (musicEnabled)
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{
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alSourceStop(currentMusic.source);
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EmptyMusicStream(); // Empty music buffers
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alDeleteSources(1, ¤tMusic.source);
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alDeleteBuffers(2, currentMusic.buffers);
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stb_vorbis_close(currentMusic.stream);
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}
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musicEnabled = false;
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}
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@ -514,9 +514,9 @@ void PauseMusicStream()
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bool MusicIsPlaying()
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{
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ALenum state;
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alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
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|
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return (state == AL_PLAYING);
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}
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@ -530,7 +530,7 @@ void SetMusicVolume(float volume)
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float GetMusicTimeLength()
|
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{
|
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float totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
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return totalSeconds;
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}
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@ -538,11 +538,11 @@ float GetMusicTimeLength()
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float GetMusicTimePlayed()
|
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{
|
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int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
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int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft;
|
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float secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels);
|
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return secondsPlayed;
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}
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@ -553,30 +553,30 @@ float GetMusicTimePlayed()
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// Fill music buffers with new data from music stream
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static bool BufferMusicStream(ALuint buffer)
|
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{
|
||||
short pcm[MUSIC_BUFFER_SIZE];
|
||||
|
||||
int size = 0; // Total size of data steamed (in bytes)
|
||||
int streamedBytes = 0; // Bytes of data obtained in one samples get
|
||||
|
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short pcm[MUSIC_BUFFER_SIZE];
|
||||
|
||||
int size = 0; // Total size of data steamed (in bytes)
|
||||
int streamedBytes = 0; // Bytes of data obtained in one samples get
|
||||
|
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bool active = true; // We can get more data from stream (not finished)
|
||||
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||||
|
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if (musicEnabled)
|
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{
|
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while (size < MUSIC_BUFFER_SIZE)
|
||||
{
|
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streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
|
||||
|
||||
|
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if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
|
||||
else break;
|
||||
}
|
||||
|
||||
|
||||
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
|
||||
}
|
||||
|
||||
if (size > 0)
|
||||
|
||||
if (size > 0)
|
||||
{
|
||||
alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
|
||||
|
||||
|
||||
currentMusic.totalSamplesLeft -= size;
|
||||
}
|
||||
else
|
||||
@ -585,21 +585,21 @@ static bool BufferMusicStream(ALuint buffer)
|
||||
TraceLog(WARNING, "No more data obtained from stream");
|
||||
}
|
||||
|
||||
return active;
|
||||
return active;
|
||||
}
|
||||
|
||||
// Empty music buffers
|
||||
static void EmptyMusicStream()
|
||||
{
|
||||
ALuint buffer = 0;
|
||||
ALuint buffer = 0;
|
||||
int queued = 0;
|
||||
|
||||
|
||||
alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued);
|
||||
|
||||
|
||||
while(queued > 0)
|
||||
{
|
||||
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
|
||||
|
||||
|
||||
queued--;
|
||||
}
|
||||
}
|
||||
@ -610,12 +610,12 @@ extern void UpdateMusicStream()
|
||||
ALuint buffer = 0;
|
||||
ALint processed = 0;
|
||||
bool active = true;
|
||||
|
||||
|
||||
if (musicEnabled)
|
||||
{
|
||||
// Get the number of already processed buffers (if any)
|
||||
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
|
||||
|
||||
|
||||
while (processed > 0)
|
||||
{
|
||||
// Recover processed buffer for refill
|
||||
@ -623,32 +623,32 @@ extern void UpdateMusicStream()
|
||||
|
||||
// Refill buffer
|
||||
active = BufferMusicStream(buffer);
|
||||
|
||||
|
||||
// If no more data to stream, restart music (if loop)
|
||||
if ((!active) && (currentMusic.loop))
|
||||
if ((!active) && (currentMusic.loop))
|
||||
{
|
||||
if (currentMusic.loop)
|
||||
{
|
||||
stb_vorbis_seek_start(currentMusic.stream);
|
||||
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
|
||||
|
||||
|
||||
active = BufferMusicStream(buffer);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Add refilled buffer to queue again... don't let the music stop!
|
||||
alSourceQueueBuffers(currentMusic.source, 1, &buffer);
|
||||
|
||||
|
||||
if(alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Ogg playing, error buffering data...");
|
||||
|
||||
|
||||
processed--;
|
||||
}
|
||||
|
||||
|
||||
ALenum state;
|
||||
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
|
||||
|
||||
|
||||
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
|
||||
|
||||
|
||||
if (!active) StopMusicStream();
|
||||
}
|
||||
}
|
||||
@ -678,16 +678,16 @@ static Wave LoadWAV(const char *fileName)
|
||||
char subChunkID[4];
|
||||
long subChunkSize;
|
||||
} WaveData;
|
||||
|
||||
|
||||
RiffHeader riffHeader;
|
||||
WaveFormat waveFormat;
|
||||
WaveData waveData;
|
||||
|
||||
|
||||
Wave wave;
|
||||
FILE *wavFile;
|
||||
|
||||
|
||||
wavFile = fopen(fileName, "rb");
|
||||
|
||||
|
||||
if (!wavFile)
|
||||
{
|
||||
TraceLog(WARNING, "[%s] Could not open WAV file", fileName);
|
||||
@ -696,7 +696,7 @@ static Wave LoadWAV(const char *fileName)
|
||||
{
|
||||
// Read in the first chunk into the struct
|
||||
fread(&riffHeader, sizeof(RiffHeader), 1, wavFile);
|
||||
|
||||
|
||||
// Check for RIFF and WAVE tags
|
||||
if (((riffHeader.chunkID[0] != 'R') || (riffHeader.chunkID[1] != 'I') || (riffHeader.chunkID[2] != 'F') || (riffHeader.chunkID[3] != 'F')) ||
|
||||
((riffHeader.format[0] != 'W') || (riffHeader.format[1] != 'A') || (riffHeader.format[2] != 'V') || (riffHeader.format[3] != 'E')))
|
||||
@ -707,7 +707,7 @@ static Wave LoadWAV(const char *fileName)
|
||||
{
|
||||
// Read in the 2nd chunk for the wave info
|
||||
fread(&waveFormat, sizeof(WaveFormat), 1, wavFile);
|
||||
|
||||
|
||||
// Check for fmt tag
|
||||
if ((waveFormat.subChunkID[0] != 'f') || (waveFormat.subChunkID[1] != 'm') ||
|
||||
(waveFormat.subChunkID[2] != 't') || (waveFormat.subChunkID[3] != ' '))
|
||||
@ -718,10 +718,10 @@ static Wave LoadWAV(const char *fileName)
|
||||
{
|
||||
// Check for extra parameters;
|
||||
if (waveFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
|
||||
|
||||
|
||||
// Read in the the last byte of data before the sound file
|
||||
fread(&waveData, sizeof(WaveData), 1, wavFile);
|
||||
|
||||
|
||||
// Check for data tag
|
||||
if ((waveData.subChunkID[0] != 'd') || (waveData.subChunkID[1] != 'a') ||
|
||||
(waveData.subChunkID[2] != 't') || (waveData.subChunkID[3] != 'a'))
|
||||
@ -731,17 +731,17 @@ static Wave LoadWAV(const char *fileName)
|
||||
else
|
||||
{
|
||||
// Allocate memory for data
|
||||
wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
|
||||
|
||||
wave.data = (unsigned char *)malloc(sizeof(unsigned char) * waveData.subChunkSize);
|
||||
|
||||
// Read in the sound data into the soundData variable
|
||||
fread(wave.data, waveData.subChunkSize, 1, wavFile);
|
||||
|
||||
|
||||
// Now we set the variables that we need later
|
||||
wave.dataSize = waveData.subChunkSize;
|
||||
wave.sampleRate = waveFormat.sampleRate;
|
||||
wave.channels = waveFormat.numChannels;
|
||||
wave.bitsPerSample = waveFormat.bitsPerSample;
|
||||
|
||||
|
||||
TraceLog(INFO, "[%s] Wave file loaded successfully", fileName);
|
||||
}
|
||||
}
|
||||
@ -749,7 +749,7 @@ static Wave LoadWAV(const char *fileName)
|
||||
|
||||
fclose(wavFile);
|
||||
}
|
||||
|
||||
|
||||
return wave;
|
||||
}
|
||||
|
||||
@ -757,42 +757,42 @@ static Wave LoadWAV(const char *fileName)
|
||||
static Wave LoadOGG(char *fileName)
|
||||
{
|
||||
Wave wave;
|
||||
|
||||
|
||||
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
|
||||
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
|
||||
|
||||
|
||||
wave.sampleRate = info.sample_rate;
|
||||
wave.bitsPerSample = 16;
|
||||
wave.channels = info.channels;
|
||||
|
||||
|
||||
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
|
||||
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
|
||||
|
||||
int totalSamplesLength = (stb_vorbis_stream_length_in_samples(oggFile) * info.channels);
|
||||
|
||||
|
||||
wave.dataSize = totalSamplesLength*sizeof(short); // Size must be in bytes
|
||||
|
||||
|
||||
TraceLog(DEBUG, "[%s] Samples length: %i", fileName, totalSamplesLength);
|
||||
|
||||
|
||||
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
|
||||
|
||||
|
||||
TraceLog(DEBUG, "[%s] Total seconds: %f", fileName, totalSeconds);
|
||||
|
||||
|
||||
if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
|
||||
|
||||
|
||||
int totalSamples = totalSeconds*info.sample_rate*info.channels;
|
||||
|
||||
|
||||
TraceLog(DEBUG, "[%s] Total samples calculated: %i", fileName, totalSamples);
|
||||
|
||||
//short *data
|
||||
|
||||
//short *data
|
||||
wave.data = malloc(sizeof(short)*totalSamplesLength);
|
||||
|
||||
int samplesObtained = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, wave.data, totalSamplesLength);
|
||||
|
||||
|
||||
TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained);
|
||||
|
||||
stb_vorbis_close(oggFile);
|
||||
|
||||
|
||||
return wave;
|
||||
}
|
||||
|
||||
|
||||
Reference in New Issue
Block a user