709 lines
20 KiB
C++
709 lines
20 KiB
C++
/*
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This file is part of Telegram Desktop,
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the official desktop version of Telegram messaging app, see https://telegram.org
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Telegram Desktop is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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It is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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Full license: https://github.com/telegramdesktop/tdesktop/blob/master/LICENSE
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Copyright (c) 2014 John Preston, https://desktop.telegram.org
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*/
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#include "stdafx.h"
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#include "audio.h"
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#include <AL/al.h>
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#include <AL/alc.h>
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#include <opusfile.h>
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#include <ogg/ogg.h>
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namespace {
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ALCdevice *audioDevice = 0;
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ALCcontext *audioContext = 0;
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ALuint notifySource = 0;
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ALuint notifyBuffer = 0;
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QMutex voicemsgsMutex;
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VoiceMessages *voicemsgs = 0;
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}
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bool _checkALCError() {
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ALenum errCode;
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if ((errCode = alcGetError(audioDevice)) != ALC_NO_ERROR) {
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LOG(("Audio Error: (alc) %1").arg((const char *)alcGetString(audioDevice, errCode)));
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return false;
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}
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return true;
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}
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bool _checkALError() {
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ALenum errCode;
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if ((errCode = alGetError()) != AL_NO_ERROR) {
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LOG(("Audio Error: (al) %1").arg((const char *)alGetString(errCode)));
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return false;
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}
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return true;
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}
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void audioInit() {
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if (audioDevice) return;
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audioDevice = alcOpenDevice(NULL);
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if (!audioDevice) {
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LOG(("Audio Error: default sound device not present."));
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return;
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}
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ALCint attributes[] = { ALC_STEREO_SOURCES, 8, 0 };
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audioContext = alcCreateContext(audioDevice, attributes);
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alcMakeContextCurrent(audioContext);
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if (!_checkALCError()) return audioFinish();
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ALfloat v[] = { 0.f, 0.f, -1.f, 0.f, 1.f, 0.f };
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alListener3f(AL_POSITION, 0.f, 0.f, 0.f);
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alListener3f(AL_VELOCITY, 0.f, 0.f, 0.f);
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alListenerfv(AL_ORIENTATION, v);
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alDistanceModel(AL_NONE);
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alGenSources(1, ¬ifySource);
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alSourcef(notifySource, AL_PITCH, 1.f);
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alSourcef(notifySource, AL_GAIN, 1.f);
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alSource3f(notifySource, AL_POSITION, 0, 0, 0);
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alSource3f(notifySource, AL_VELOCITY, 0, 0, 0);
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alSourcei(notifySource, AL_LOOPING, 0);
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alGenBuffers(1, ¬ifyBuffer);
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if (!_checkALError()) return audioFinish();
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QFile notify(st::newMsgSound);
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if (!notify.open(QIODevice::ReadOnly)) return audioFinish();
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QByteArray blob = notify.readAll();
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const char *data = blob.constData();
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if (blob.size() < 44) return audioFinish();
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if (*((const uint32*)(data + 0)) != 0x46464952) return audioFinish(); // ChunkID - "RIFF"
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if (*((const uint32*)(data + 4)) != uint32(blob.size() - 8)) return audioFinish(); // ChunkSize
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if (*((const uint32*)(data + 8)) != 0x45564157) return audioFinish(); // Format - "WAVE"
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if (*((const uint32*)(data + 12)) != 0x20746d66) return audioFinish(); // Subchunk1ID - "fmt "
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uint32 subchunk1Size = *((const uint32*)(data + 16)), extra = subchunk1Size - 16;
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if (subchunk1Size < 16 || (extra && extra < 2)) return audioFinish();
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if (*((const uint16*)(data + 20)) != 1) return audioFinish(); // AudioFormat - PCM (1)
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uint16 numChannels = *((const uint16*)(data + 22));
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if (numChannels != 1 && numChannels != 2) return audioFinish();
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uint32 sampleRate = *((const uint32*)(data + 24));
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uint32 byteRate = *((const uint32*)(data + 28));
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uint16 blockAlign = *((const uint16*)(data + 32));
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uint16 bitsPerSample = *((const uint16*)(data + 34));
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if (bitsPerSample % 8) return audioFinish();
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uint16 bytesPerSample = bitsPerSample / 8;
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if (bytesPerSample != 1 && bytesPerSample != 2) return audioFinish();
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if (blockAlign != numChannels * bytesPerSample) return audioFinish();
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if (byteRate != sampleRate * blockAlign) return audioFinish();
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if (extra) {
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uint16 extraSize = *((const uint16*)(data + 36));
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if (uint32(extraSize + 2) != extra) return audioFinish();
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if (uint32(blob.size()) < 44 + extra) return audioFinish();
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}
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if (*((const uint32*)(data + extra + 36)) != 0x61746164) return audioFinish(); // Subchunk2ID - "data"
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uint32 subchunk2Size = *((const uint32*)(data + extra + 40));
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if (subchunk2Size % (numChannels * bytesPerSample)) return audioFinish();
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uint32 numSamples = subchunk2Size / (numChannels * bytesPerSample);
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if (uint32(blob.size()) < 44 + extra + subchunk2Size) return audioFinish();
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data += 44 + extra;
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ALenum format = 0;
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switch (bytesPerSample) {
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case 1:
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switch (numChannels) {
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case 1: format = AL_FORMAT_MONO8; break;
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case 2: format = AL_FORMAT_STEREO8; break;
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}
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break;
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case 2:
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switch (numChannels) {
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case 1: format = AL_FORMAT_MONO16; break;
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case 2: format = AL_FORMAT_STEREO16; break;
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}
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break;
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}
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if (!format) return audioFinish();
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alBufferData(notifyBuffer, format, data, subchunk2Size, sampleRate);
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alSourcei(notifySource, AL_BUFFER, notifyBuffer);
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if (!_checkALError()) return audioFinish();
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voicemsgs = new VoiceMessages();
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}
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bool audioWorks() {
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return !!voicemsgs;
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}
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void audioPlayNotify() {
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if (!audioWorks()) return;
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alSourcePlay(notifySource);
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}
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void audioFinish() {
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if (voicemsgs) {
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delete voicemsgs;
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}
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alSourceStop(notifySource);
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if (alIsBuffer(notifyBuffer)) {
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alDeleteBuffers(1, ¬ifyBuffer);
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notifyBuffer = 0;
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}
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if (alIsSource(notifySource)) {
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alDeleteSources(1, ¬ifySource);
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notifySource = 0;
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}
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if (audioContext) {
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alcMakeContextCurrent(NULL);
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alcDestroyContext(audioContext);
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audioContext = 0;
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}
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if (audioDevice) {
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alcCloseDevice(audioDevice);
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audioDevice = 0;
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}
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}
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VoiceMessages::VoiceMessages() : _current(0),
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_fader(new VoiceMessagesFader(&_faderThread)), _loader(new VoiceMessagesLoader(&_loaderThread)) {
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connect(this, SIGNAL(faderOnTimer()), _fader, SLOT(onTimer()));
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connect(this, SIGNAL(loaderOnStart(AudioData*)), _loader, SLOT(onStart(AudioData*)));
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connect(this, SIGNAL(loaderOnCancel(AudioData*)), _loader, SLOT(onCancel(AudioData*)));
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connect(&_faderThread, SIGNAL(started()), _fader, SLOT(onInit()));
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connect(&_loaderThread, SIGNAL(started()), _loader, SLOT(onInit()));
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connect(&_faderThread, SIGNAL(finished()), _fader, SLOT(deleteLater()));
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connect(&_loaderThread, SIGNAL(finished()), _loader, SLOT(deleteLater()));
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connect(_loader, SIGNAL(needToCheck()), _fader, SLOT(onTimer()));
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connect(_loader, SIGNAL(error(AudioData*)), this, SLOT(onError(AudioData*)));
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connect(_fader, SIGNAL(needToPreload(AudioData*)), _loader, SLOT(onLoad(AudioData*)));
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connect(_fader, SIGNAL(playPositionUpdated(AudioData*)), this, SIGNAL(updated(AudioData*)));
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connect(_fader, SIGNAL(audioStopped(AudioData*)), this, SIGNAL(stopped(AudioData*)));
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connect(_fader, SIGNAL(error(AudioData*)), this, SLOT(onError(AudioData*)));
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_loaderThread.start();
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_faderThread.start();
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}
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VoiceMessages::~VoiceMessages() {
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{
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QMutexLocker lock(&voicemsgsMutex);
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voicemsgs = 0;
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}
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for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
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alSourceStop(_data[i].source);
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if (alIsBuffer(_data[i].buffers[0])) {
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alDeleteBuffers(3, _data[i].buffers);
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for (int32 j = 0; j < 3; ++j) {
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_data[i].buffers[j] = _data[i].samplesCount[j] = 0;
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}
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}
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if (alIsSource(_data[i].source)) {
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alDeleteSources(1, &_data[i].source);
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_data[i].source = 0;
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}
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}
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_faderThread.quit();
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_loaderThread.quit();
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_faderThread.wait();
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_loaderThread.wait();
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}
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void VoiceMessages::onError(AudioData *audio) {
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emit stopped(audio);
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}
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bool VoiceMessages::updateCurrentStarted(int32 pos) {
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if (pos < 0) {
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if (alIsSource(_data[_current].source)) {
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alGetSourcei(_data[_current].source, AL_SAMPLE_OFFSET, &pos);
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} else {
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pos = 0;
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}
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}
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if (!_checkALError()) {
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_data[_current].state = VoiceMessageStopped;
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onError(_data[_current].audio);
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return false;
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}
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_data[_current].started = _data[_current].position = pos + _data[_current].skipStart;
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return true;
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}
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void VoiceMessages::play(AudioData *audio) {
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QMutexLocker lock(&voicemsgsMutex);
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bool startNow = true;
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if (_data[_current].audio != audio) {
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switch (_data[_current].state) {
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case VoiceMessageStarting:
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case VoiceMessageResuming:
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case VoiceMessagePlaying:
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_data[_current].state = VoiceMessageFinishing;
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updateCurrentStarted();
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startNow = false;
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break;
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case VoiceMessagePausing: _data[_current].state = VoiceMessageFinishing; startNow = false; break;
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case VoiceMessagePaused: _data[_current].state = VoiceMessageStopped; break;
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}
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if (_data[_current].audio) {
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emit loaderOnCancel(_data[_current].audio);
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emit faderOnTimer();
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}
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}
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int32 index = 0;
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for (; index < AudioVoiceMsgSimultaneously; ++index) {
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if (_data[index].audio == audio) {
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_current = index;
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break;
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}
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}
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if (index == AudioVoiceMsgSimultaneously && ++_current >= AudioVoiceMsgSimultaneously) {
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_current -= AudioVoiceMsgSimultaneously;
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}
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_data[_current].audio = audio;
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_data[_current].fname = audio->already(true);
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_data[_current].data = audio->data;
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if (_data[_current].fname.isEmpty() && _data[_current].data.isEmpty()) {
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_data[_current].state = VoiceMessageStopped;
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onError(audio);
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} else if (updateCurrentStarted(0)) {
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_data[_current].state = startNow ? VoiceMessagePlaying : VoiceMessageStarting;
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_data[_current].loading = true;
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emit loaderOnStart(audio);
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}
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}
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void VoiceMessages::pauseresume() {
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QMutexLocker lock(&voicemsgsMutex);
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switch (_data[_current].state) {
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case VoiceMessagePausing:
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case VoiceMessagePaused:
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if (_data[_current].state == VoiceMessagePaused) {
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updateCurrentStarted();
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}
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_data[_current].state = VoiceMessageResuming;
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alSourcePlay(_data[_current].source);
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break;
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case VoiceMessageStarting:
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case VoiceMessageResuming:
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case VoiceMessagePlaying:
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_data[_current].state = VoiceMessagePausing;
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updateCurrentStarted();
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break;
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case VoiceMessageFinishing: _data[_current].state = VoiceMessagePausing; break;
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}
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emit faderOnTimer();
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}
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void VoiceMessages::currentState(AudioData **audio, VoiceMessageState *state, int64 *position, int64 *duration) {
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QMutexLocker lock(&voicemsgsMutex);
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if (audio) *audio = _data[_current].audio;
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if (state) *state = _data[_current].state;
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if (position) *position = _data[_current].position;
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if (duration) *duration = _data[_current].duration;
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}
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VoiceMessages *audioVoice() {
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return voicemsgs;
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}
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VoiceMessagesFader::VoiceMessagesFader(QThread *thread) : _timer(this) {
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moveToThread(thread);
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_timer.moveToThread(thread);
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_timer.setSingleShot(true);
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connect(&_timer, SIGNAL(timeout()), this, SLOT(onTimer()));
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}
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void VoiceMessagesFader::onInit() {
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}
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void VoiceMessagesFader::onTimer() {
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bool hasFading = false, hasPlaying = false;
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QMutexLocker lock(&voicemsgsMutex);
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VoiceMessages *voice = audioVoice();
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if (!voice) return;
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for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
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VoiceMessages::Msg &m(voice->_data[i]);
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if (m.state == VoiceMessageStopped || m.state == VoiceMessagePaused || !m.source) continue;
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bool playing = false, fading = false;
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ALint pos = 0;
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ALint state = AL_INITIAL;
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alGetSourcei(m.source, AL_SAMPLE_OFFSET, &pos);
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alGetSourcei(m.source, AL_SOURCE_STATE, &state);
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if (!_checkALError()) {
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m.state = VoiceMessageStopped;
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emit error(m.audio);
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} else {
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switch (m.state) {
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case VoiceMessageFinishing:
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case VoiceMessagePausing:
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case VoiceMessageStarting:
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case VoiceMessageResuming:
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fading = true;
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break;
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case VoiceMessagePlaying:
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playing = true;
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break;
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}
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if (fading && (state == AL_PLAYING || !m.loading)) {
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if (state != AL_PLAYING) {
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fading = false;
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if (m.source) {
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alSourcef(m.source, AL_GAIN, 1);
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alSourceStop(m.source);
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}
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m.state = VoiceMessageStopped;
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emit audioStopped(m.audio);
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} else if (1000 * (pos + m.skipStart - m.started) >= AudioFadeDuration * AudioVoiceMsgFrequency) {
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fading = false;
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alSourcef(m.source, AL_GAIN, 1);
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switch (m.state) {
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case VoiceMessageFinishing: alSourceStop(m.source); m.state = VoiceMessageStopped; break;
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case VoiceMessagePausing: alSourcePause(m.source); m.state = VoiceMessagePaused; break;
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case VoiceMessageStarting:
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case VoiceMessageResuming:
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m.state = VoiceMessagePlaying;
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playing = true;
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break;
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}
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} else {
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float64 newGain = 1000. * (pos + m.skipStart - m.started) / (AudioFadeDuration * AudioVoiceMsgFrequency);
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if (m.state == VoiceMessagePausing || m.state == VoiceMessageFinishing) {
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newGain = 1. - newGain;
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}
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if (newGain < 0) {
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int a = 0, b;
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b = a;
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}
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alSourcef(m.source, AL_GAIN, newGain);
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LOG(("Now volume is: %1").arg(newGain));
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}
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} else if (playing && (state == AL_PLAYING || !m.loading)) {
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if (state != AL_PLAYING) {
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playing = false;
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if (m.source) {
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alSourceStop(m.source);
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alSourcef(m.source, AL_GAIN, 1);
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}
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m.state = VoiceMessageStopped;
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emit audioStopped(m.audio);
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}
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}
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if (pos + m.skipStart - m.position >= AudioCheckPositionDelta) {
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m.position = pos + m.skipStart;
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emit playPositionUpdated(m.audio);
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}
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if (!m.loading && m.skipEnd > 0 && m.position + AudioPreloadSamples + m.skipEnd > m.duration) {
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m.loading = true;
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emit needToPreload(m.audio);
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}
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if (playing) hasPlaying = true;
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if (fading) hasFading = true;
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}
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}
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if (hasFading) {
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_timer.start(AudioFadeTimeout);
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} else if (hasPlaying) {
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_timer.start(AudioCheckPositionTimeout);
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}
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}
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struct VoiceMessagesLoader::Loader {
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QString fname;
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QByteArray data;
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OggOpusFile *file;
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ogg_int64_t pcm_offset;
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ogg_int64_t pcm_print_offset;
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int prev_li;
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Loader() : file(0), pcm_offset(0), pcm_print_offset(0), prev_li(-1) {
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}
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};
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VoiceMessagesLoader::VoiceMessagesLoader(QThread *thread) {
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moveToThread(thread);
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}
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VoiceMessagesLoader::~VoiceMessagesLoader() {
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for (Loaders::iterator i = _loaders.begin(), e = _loaders.end(); i != e; ++i) {
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delete i.value();
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}
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_loaders.clear();
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}
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void VoiceMessagesLoader::onInit() {
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}
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void VoiceMessagesLoader::onStart(AudioData *audio) {
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Loaders::iterator i = _loaders.find(audio);
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if (i != _loaders.end()) {
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delete (*i);
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_loaders.erase(i);
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}
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onLoad(audio);
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}
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void VoiceMessagesLoader::loadError(Loaders::iterator i) {
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emit error(i.key());
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delete (*i);
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_loaders.erase(i);
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}
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void VoiceMessagesLoader::onLoad(AudioData *audio) {
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bool started = false;
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int32 audioindex = -1;
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Loader *l = 0;
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Loaders::iterator j = _loaders.end();
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{
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QMutexLocker lock(&voicemsgsMutex);
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VoiceMessages *voice = audioVoice();
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if (!voice) return;
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for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
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VoiceMessages::Msg &m(voice->_data[i]);
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if (m.audio != audio || !m.loading) continue;
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audioindex = i;
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j = _loaders.find(audio);
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if (j != _loaders.end() && (j.value()->fname != m.fname || j.value()->data.size() != m.data.size())) {
|
|
delete j.value();
|
|
_loaders.erase(j);
|
|
j = _loaders.end();
|
|
}
|
|
if (j == _loaders.end()) {
|
|
l = (j = _loaders.insert(audio, new Loader())).value();
|
|
l->fname = m.fname;
|
|
l->data = m.data;
|
|
|
|
int ret;
|
|
if (m.data.isEmpty()) {
|
|
l->file = op_open_file(m.fname.toUtf8().constData(), &ret);
|
|
} else {
|
|
l->file = op_open_memory((const unsigned char*)m.data.constData(), m.data.size(), &ret);
|
|
}
|
|
if (!l->file) {
|
|
LOG(("Audio Error: op_open_file failed for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(ret));
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
ogg_int64_t duration = op_pcm_total(l->file, -1);
|
|
if (duration < 0) {
|
|
LOG(("Audio Error: op_pcm_total failed to get full duration for '%1', data size '%2', error code %3").arg(m.fname).arg(m.data.size()).arg(duration));
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
m.duration = duration;
|
|
m.skipStart = 0;
|
|
m.skipEnd = duration;
|
|
m.position = 0;
|
|
m.started = 0;
|
|
started = true;
|
|
} else {
|
|
if (!m.skipEnd) continue;
|
|
l = j.value();
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (j == _loaders.end()) {
|
|
LOG(("Audio Error: trying to load part of audio, that is not playing at the moment"));
|
|
emit error(audio);
|
|
return;
|
|
}
|
|
if (started) {
|
|
l->pcm_offset = op_pcm_tell(l->file);
|
|
l->pcm_print_offset = l->pcm_offset - AudioVoiceMsgFrequency;
|
|
}
|
|
|
|
bool finished = false;
|
|
DEBUG_LOG(("Audio Info: reading buffer for file '%1', data size '%2', current pcm_offset %3").arg(l->fname).arg(l->data.size()).arg(l->pcm_offset));
|
|
|
|
QByteArray result;
|
|
int64 samplesAdded = 0;
|
|
while (result.size() < AudioVoiceMsgBufferSize) {
|
|
opus_int16 pcm[AudioVoiceMsgFrequency * AudioVoiceMsgChannels];
|
|
|
|
int ret = op_read_stereo(l->file, pcm, sizeof(pcm) / sizeof(*pcm));
|
|
if (ret < 0) {
|
|
{
|
|
QMutexLocker lock(&voicemsgsMutex);
|
|
VoiceMessages *voice = audioVoice();
|
|
if (voice) {
|
|
VoiceMessages::Msg &m(voice->_data[audioindex]);
|
|
if (m.audio == audio) {
|
|
m.state = VoiceMessageStopped;
|
|
}
|
|
}
|
|
}
|
|
LOG(("Audio Error: op_read_stereo failed, error code %1").arg(ret));
|
|
return loadError(j);
|
|
}
|
|
|
|
int li = op_current_link(l->file);
|
|
if (li != l->prev_li) {
|
|
const OpusHead *head = op_head(l->file, li);
|
|
const OpusTags *tags = op_tags(l->file, li);
|
|
for (int32 ci = 0; ci < tags->comments; ++ci) {
|
|
const char *comment = tags->user_comments[ci];
|
|
if (opus_tagncompare("METADATA_BLOCK_PICTURE", 22, comment) == 0) {
|
|
OpusPictureTag pic;
|
|
int err = opus_picture_tag_parse(&pic, comment);
|
|
if (err >= 0) {
|
|
opus_picture_tag_clear(&pic);
|
|
}
|
|
}
|
|
}
|
|
if (!op_seekable(l->file)) {
|
|
l->pcm_offset = op_pcm_tell(l->file) - ret;
|
|
}
|
|
}
|
|
if (li != l->prev_li || l->pcm_offset >= l->pcm_print_offset + AudioVoiceMsgFrequency) {
|
|
l->pcm_print_offset = l->pcm_offset;
|
|
}
|
|
l->pcm_offset = op_pcm_tell(l->file);
|
|
|
|
if (!ret) {
|
|
DEBUG_LOG(("Audio Info: read completed"));
|
|
finished = true;
|
|
break;
|
|
}
|
|
result.append((const char*)pcm, sizeof(*pcm) * ret * AudioVoiceMsgChannels);
|
|
l->prev_li = li;
|
|
samplesAdded += ret;
|
|
|
|
{
|
|
QMutexLocker lock(&voicemsgsMutex);
|
|
VoiceMessages *voice = audioVoice();
|
|
if (!voice) return;
|
|
|
|
VoiceMessages::Msg &m(voice->_data[audioindex]);
|
|
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
|
|
LOG(("Audio Error: playing changed while loading"));
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
}
|
|
}
|
|
|
|
QMutexLocker lock(&voicemsgsMutex);
|
|
VoiceMessages *voice = audioVoice();
|
|
if (!voice) return;
|
|
|
|
VoiceMessages::Msg &m(voice->_data[audioindex]);
|
|
if (m.audio != audio || !m.loading || m.fname != l->fname || m.data.size() != l->data.size()) {
|
|
LOG(("Audio Error: playing changed while loading"));
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
|
|
if (started) {
|
|
if (m.source) {
|
|
alSourceStop(m.source);
|
|
for (int32 i = 0; i < 3; ++i) {
|
|
if (m.samplesCount[i]) {
|
|
alSourceUnqueueBuffers(m.source, 1, m.buffers + i);
|
|
m.samplesCount[i] = 0;
|
|
}
|
|
}
|
|
m.nextBuffer = 0;
|
|
}
|
|
}
|
|
if (samplesAdded) {
|
|
if (!m.source) {
|
|
alGenSources(1, &m.source);
|
|
alSourcef(m.source, AL_PITCH, 1.f);
|
|
alSourcef(m.source, AL_GAIN, 1.f);
|
|
alSource3f(m.source, AL_POSITION, 0, 0, 0);
|
|
alSource3f(m.source, AL_VELOCITY, 0, 0, 0);
|
|
alSourcei(m.source, AL_LOOPING, 0);
|
|
}
|
|
if (!m.buffers[m.nextBuffer]) alGenBuffers(3, m.buffers);
|
|
if (!_checkALError()) {
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
|
|
if (m.samplesCount[m.nextBuffer]) {
|
|
alSourceUnqueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
|
|
m.skipStart += m.samplesCount[m.nextBuffer];
|
|
}
|
|
|
|
m.samplesCount[m.nextBuffer] = samplesAdded;
|
|
alBufferData(m.buffers[m.nextBuffer], AL_FORMAT_STEREO16, result.constData(), result.size(), AudioVoiceMsgFrequency);
|
|
alSourceQueueBuffers(m.source, 1, m.buffers + m.nextBuffer);
|
|
m.skipEnd -= samplesAdded;
|
|
|
|
m.nextBuffer = (m.nextBuffer + 1) % 3;
|
|
|
|
if (!_checkALError()) {
|
|
m.state = VoiceMessageStopped;
|
|
return loadError(j);
|
|
}
|
|
} else {
|
|
finished = true;
|
|
}
|
|
if (finished) {
|
|
m.skipEnd = 0;
|
|
m.duration = m.skipStart + m.samplesCount[0] + m.samplesCount[1] + m.samplesCount[2];
|
|
}
|
|
m.loading = false;
|
|
if (m.state == VoiceMessageResuming || m.state == VoiceMessagePlaying || m.state == VoiceMessageStarting) {
|
|
ALint state = AL_INITIAL;
|
|
alGetSourcei(m.source, AL_SOURCE_STATE, &state);
|
|
if (_checkALError()) {
|
|
if (state != AL_PLAYING) {
|
|
alSourcePlay(m.source);
|
|
emit needToCheck();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void VoiceMessagesLoader::onCancel(AudioData *audio) {
|
|
Loaders::iterator i = _loaders.find(audio);
|
|
if (i != _loaders.end()) {
|
|
delete (*i);
|
|
_loaders.erase(i);
|
|
}
|
|
|
|
QMutexLocker lock(&voicemsgsMutex);
|
|
VoiceMessages *voice = audioVoice();
|
|
if (!voice) return;
|
|
|
|
for (int32 i = 0; i < AudioVoiceMsgSimultaneously; ++i) {
|
|
VoiceMessages::Msg &m(voice->_data[i]);
|
|
if (m.audio == audio) {
|
|
m.loading = false;
|
|
}
|
|
}
|
|
}
|