/* ** oalsound.cpp ** System interface for sound; uses OpenAL ** **--------------------------------------------------------------------------- ** Copyright 2008-2010 Chris Robinson ** All rights reserved. ** ** Redistribution and use in source and binary forms, with or without ** modification, are permitted provided that the following conditions ** are met: ** ** 1. Redistributions of source code must retain the above copyright ** notice, this list of conditions and the following disclaimer. ** 2. Redistributions in binary form must reproduce the above copyright ** notice, this list of conditions and the following disclaimer in the ** documentation and/or other materials provided with the distribution. ** 3. The name of the author may not be used to endorse or promote products ** derived from this software without specific prior written permission. ** ** THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR ** IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES ** OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. ** IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, ** INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT ** NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, ** DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY ** THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT ** (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF ** THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. **--------------------------------------------------------------------------- ** */ #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include #define USE_WINDOWS_DWORD #endif #include "doomstat.h" #include "templates.h" #include "oalsound.h" #include "c_cvars.h" #include "c_dispatch.h" #include "i_system.h" #include "v_text.h" #include "gi.h" #include "actor.h" #include "r_state.h" #include "w_wad.h" #include "i_music.h" #include "i_musicinterns.h" #include "tempfiles.h" CVAR (String, snd_aldevice, "Default", CVAR_ARCHIVE|CVAR_GLOBALCONFIG) CVAR (Bool, snd_efx, true, CVAR_ARCHIVE|CVAR_GLOBALCONFIG) CUSTOM_CVAR (Float, snd_waterabsorption, 10.0f, CVAR_ARCHIVE|CVAR_GLOBALCONFIG) { if(*self < 0.0f) self = 0.0f; else if(*self > 10.0f) self = 10.0f; } void I_BuildALDeviceList(FOptionValues *opt) { opt->mValues.Resize(1); opt->mValues[0].TextValue = "Default"; opt->mValues[0].Text = "Default"; #ifndef NO_OPENAL const ALCchar *names = (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") ? alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER) : alcGetString(NULL, ALC_DEVICE_SPECIFIER)); if(!names) Printf("Failed to get device list: %s\n", alcGetString(NULL, alcGetError(NULL))); else while(*names) { unsigned int i = opt->mValues.Reserve(1); opt->mValues[i].TextValue = names; opt->mValues[i].Text = names; names += strlen(names)+1; } #endif } #ifndef NO_OPENAL #include #include #include #include EXTERN_CVAR (Int, snd_channels) EXTERN_CVAR (Int, snd_samplerate) EXTERN_CVAR (Bool, snd_waterreverb) EXTERN_CVAR (Bool, snd_pitched) #define foreach(type, name, vec) \ for(std::vector::iterator (name) = (vec).begin(), \ (_end_##name) = (vec).end(); \ (name) != (_end_##name);(name)++) static ALenum checkALError(const char *fn, unsigned int ln) { ALenum err = alGetError(); if(err != AL_NO_ERROR) { if(strchr(fn, '/')) fn = strrchr(fn, '/')+1; else if(strchr(fn, '\\')) fn = strrchr(fn, '\\')+1; Printf(">>>>>>>>>>>> Received AL error %s (%#x), %s:%u\n", alGetString(err), err, fn, ln); } return err; } #define getALError() checkALError(__FILE__, __LINE__) static ALCenum checkALCError(ALCdevice *device, const char *fn, unsigned int ln) { ALCenum err = alcGetError(device); if(err != ALC_NO_ERROR) { if(strchr(fn, '/')) fn = strrchr(fn, '/')+1; else if(strchr(fn, '\\')) fn = strrchr(fn, '\\')+1; Printf(">>>>>>>>>>>> Received ALC error %s (%#x), %s:%u\n", alcGetString(device, err), err, fn, ln); } return err; } #define getALCError(d) checkALCError((d), __FILE__, __LINE__) static ALsizei GetBufferLength(ALuint buffer, const char *fn, unsigned int ln) { ALint bits, channels, size; alGetBufferi(buffer, AL_BITS, &bits); alGetBufferi(buffer, AL_CHANNELS, &channels); alGetBufferi(buffer, AL_SIZE, &size); if(checkALError(fn, ln) == AL_NO_ERROR) return (ALsizei)(size / (channels * bits / 8)); return 0; } #define getBufferLength(b) GetBufferLength((b), __FILE__, __LINE__) extern ReverbContainer *ForcedEnvironment; #define PITCH_MULT (0.7937005f) /* Approx. 4 semitones lower; what Nash suggested */ #define PITCH(pitch) (snd_pitched ? (pitch)/128.f : 1.f) static float GetRolloff(const FRolloffInfo *rolloff, float distance) { if(distance <= rolloff->MinDistance) return 1.f; // Logarithmic rolloff has no max distance where it goes silent. if(rolloff->RolloffType == ROLLOFF_Log) return rolloff->MinDistance / (rolloff->MinDistance + rolloff->RolloffFactor*(distance-rolloff->MinDistance)); if(distance >= rolloff->MaxDistance) return 0.f; float volume = (rolloff->MaxDistance - distance) / (rolloff->MaxDistance - rolloff->MinDistance); if(rolloff->RolloffType == ROLLOFF_Linear) return volume; if(rolloff->RolloffType == ROLLOFF_Custom && S_SoundCurve != NULL) return S_SoundCurve[int(S_SoundCurveSize * (1.f - volume))] / 127.f; return (powf(10.f, volume) - 1.f) / 9.f; } static ALenum FormatFromDesc(int bits, int channels) { if(bits == 8) { if(channels == 1) return AL_FORMAT_MONO8; if(channels == 2) return AL_FORMAT_STEREO8; if(channels == 4) return AL_FORMAT_QUAD8; if(channels == 6) return AL_FORMAT_51CHN8; if(channels == 7) return AL_FORMAT_61CHN8; if(channels == 8) return AL_FORMAT_71CHN8; } if(bits == 16) { if(channels == 1) return AL_FORMAT_MONO16; if(channels == 2) return AL_FORMAT_STEREO16; if(channels == 4) return AL_FORMAT_QUAD16; if(channels == 6) return AL_FORMAT_51CHN16; if(channels == 7) return AL_FORMAT_61CHN16; if(channels == 8) return AL_FORMAT_71CHN16; } if(bits == 32) { if(channels == 1) return AL_FORMAT_MONO_FLOAT32; if(channels == 2) return AL_FORMAT_STEREO_FLOAT32; if(channels == 4) return AL_FORMAT_QUAD32; if(channels == 6) return AL_FORMAT_51CHN32; if(channels == 7) return AL_FORMAT_61CHN32; if(channels == 8) return AL_FORMAT_71CHN32; } return AL_NONE; } #ifdef WITH_GSTREAMER #include #include #include #include #include /* Bad GStreamer, using enums for bit fields... */ static GstMessageType operator|(const GstMessageType &a, const GstMessageType &b) { return GstMessageType((unsigned)a|(unsigned)b); } static GstSeekFlags operator|(const GstSeekFlags &a, const GstSeekFlags &b) { return GstSeekFlags((unsigned)a|(unsigned)b); } static void PrintErrMsg(const char *str, GstMessage *msg) { GError *error; gchar *debug; gst_message_parse_error(msg, &error, &debug); Printf("%s: %s\n", str, error->message); DPrintf("%s\n", debug); g_error_free(error); g_free(debug); } static GstCaps *SupportedBufferFormatCaps(int forcebits=0) { GstStructure *structure; GstCaps *caps; caps = gst_caps_new_empty(); if(alIsExtensionPresent("AL_EXT_MCFORMATS")) { static const struct { gint count; GstAudioChannelPosition pos[8]; } chans[] = { { 1, { GST_AUDIO_CHANNEL_POSITION_FRONT_MONO } }, { 2, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT } }, { 4, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT } }, { 6, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT } }, { 7, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT } }, { 8, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT } }, }; static const char *fmt32[] = { "AL_FORMAT_MONO_FLOAT32", "AL_FORMAT_STEREO_FLOAT32", "AL_FORMAT_QUAD32", "AL_FORMAT_51CHN32", "AL_FORMAT_61CHN32", "AL_FORMAT_71CHN32", NULL }; static const char *fmt16[] = { "AL_FORMAT_MONO16", "AL_FORMAT_STEREO16", "AL_FORMAT_QUAD16", "AL_FORMAT_51CHN16", "AL_FORMAT_61CHN16", "AL_FORMAT_71CHN16", NULL }; static const char *fmt8[] = { "AL_FORMAT_MONO8", "AL_FORMAT_STEREO8", "AL_FORMAT_QUAD8", "AL_FORMAT_51CHN8", "AL_FORMAT_61CHN8", "AL_FORMAT_71CHN8", NULL }; if(alIsExtensionPresent("AL_EXT_FLOAT32")) { for(size_t i = 0;fmt32[i];i++) { if(forcebits && forcebits != 32) break; ALenum val = alGetEnumValue(fmt32[i]); if(getALError() != AL_NO_ERROR || val == 0 || val == -1) continue; structure = gst_structure_new("audio/x-raw-float", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); gst_structure_set(structure, "channels", G_TYPE_INT, chans[i].count, NULL); if(chans[i].count > 2) gst_audio_set_channel_positions(structure, chans[i].pos); gst_caps_append_structure(caps, structure); } } for(size_t i = 0;fmt16[i];i++) { if(forcebits && forcebits != 16) break; ALenum val = alGetEnumValue(fmt16[i]); if(getALError() != AL_NO_ERROR || val == 0 || val == -1) continue; structure = gst_structure_new("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL); gst_structure_set(structure, "channels", G_TYPE_INT, chans[i].count, NULL); if(chans[i].count > 2) gst_audio_set_channel_positions(structure, chans[i].pos); gst_caps_append_structure(caps, structure); } for(size_t i = 0;fmt8[i];i++) { if(forcebits && forcebits != 8) break; ALenum val = alGetEnumValue(fmt8[i]); if(getALError() != AL_NO_ERROR || val == 0 || val == -1) continue; structure = gst_structure_new("audio/x-raw-int", "width", G_TYPE_INT, 8, "depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, FALSE, NULL); gst_structure_set(structure, "channels", G_TYPE_INT, chans[i].count, NULL); if(chans[i].count > 2) gst_audio_set_channel_positions(structure, chans[i].pos); gst_caps_append_structure(caps, structure); } } else { if(alIsExtensionPresent("AL_EXT_FLOAT32") && (!forcebits || forcebits == 32)) { structure = gst_structure_new("audio/x-raw-float", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); gst_caps_append_structure(caps, structure); } if(!forcebits || forcebits == 16) { structure = gst_structure_new("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); gst_caps_append_structure(caps, structure); } if(!forcebits || forcebits == 8) { structure = gst_structure_new("audio/x-raw-int", "width", G_TYPE_INT, 8, "depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, FALSE, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); gst_caps_append_structure(caps, structure); } } return caps; } class OpenALSoundStream : public SoundStream { OpenALSoundRenderer *Renderer; GstElement *gstPipeline; GstTagList *TagList; gint64 LoopPts[2]; ALuint Source; bool Playing; bool Looping; // Custom OpenAL sink; this is pretty crappy compared to the real // openalsink element, but it gets the job done static const ALsizei MaxSamplesQueued = 32768; std::vector Buffers; ALsizei SamplesQueued; ALsizei SampleRate; ALenum Format; static void sink_eos(GstAppSink *sink, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); if(!self->Playing) return; ALint state; do { g_usleep(10000); alGetSourcei(self->Source, AL_SOURCE_STATE, &state); } while(getALError() == AL_NO_ERROR && state == AL_PLAYING && self->Playing); alSourceRewind(self->Source); getALError(); } static GstFlowReturn sink_preroll(GstAppSink *sink, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); // get the buffer from appsink GstBuffer *buffer = gst_app_sink_pull_preroll(sink); if(!buffer) return GST_FLOW_ERROR; GstCaps *caps = GST_BUFFER_CAPS(buffer); gint bits = 0, channels = 0, rate = 0, i; for(i = gst_caps_get_size(caps)-1;i >= 0;i--) { GstStructure *struc = gst_caps_get_structure(caps, i); if(gst_structure_has_field(struc, "width")) gst_structure_get_int(struc, "width", &bits); if(gst_structure_has_field(struc, "channels")) gst_structure_get_int(struc, "channels", &channels); if(gst_structure_has_field(struc, "rate")) gst_structure_get_int(struc, "rate", &rate); } self->SampleRate = rate; self->Format = FormatFromDesc(bits, channels); gst_buffer_unref(buffer); if(self->Format == AL_NONE || self->SampleRate <= 0) return GST_FLOW_ERROR; return GST_FLOW_OK; } static GstFlowReturn sink_buffer(GstAppSink *sink, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); GstBuffer *buffer = gst_app_sink_pull_buffer(sink); if(!buffer) return GST_FLOW_ERROR; if(GST_BUFFER_SIZE(buffer) == 0) { gst_buffer_unref(buffer); return GST_FLOW_OK; } ALint processed, state; next_buffer: do { alGetSourcei(self->Source, AL_SOURCE_STATE, &state); alGetSourcei(self->Source, AL_BUFFERS_PROCESSED, &processed); if(getALError() != AL_NO_ERROR) { gst_buffer_unref(buffer); return GST_FLOW_ERROR; } if(processed > 0 || self->SamplesQueued < MaxSamplesQueued || state != AL_PLAYING || !self->Playing) break; g_usleep(10000); } while(1); if(!self->Playing) { gst_buffer_unref(buffer); return GST_FLOW_OK; } ALuint bufID; if(processed == 0) { alGenBuffers(1, &bufID); if(getALError() != AL_NO_ERROR) { gst_buffer_unref(buffer); return GST_FLOW_ERROR; } self->Buffers.push_back(bufID); } else while(1) { alSourceUnqueueBuffers(self->Source, 1, &bufID); if(getALError() != AL_NO_ERROR) { gst_buffer_unref(buffer); return GST_FLOW_ERROR; } self->SamplesQueued -= getBufferLength(bufID); processed--; if(self->SamplesQueued < MaxSamplesQueued) break; if(processed == 0) goto next_buffer; self->Buffers.erase(find(self->Buffers.begin(), self->Buffers.end(), bufID)); alDeleteBuffers(1, &bufID); } alBufferData(bufID, self->Format, GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer), self->SampleRate); alSourceQueueBuffers(self->Source, 1, &bufID); gst_buffer_unref(buffer); if(getALError() != AL_NO_ERROR) return GST_FLOW_ERROR; self->SamplesQueued += getBufferLength(bufID); if(state != AL_PLAYING && processed == 0) { alSourcePlay(self->Source); if(getALError() != AL_NO_ERROR) return GST_FLOW_ERROR; } return GST_FLOW_OK; } // Memory-based data source std::vector MemData; size_t MemDataPos; static void need_memdata(GstAppSrc *appsrc, guint size, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); if(self->MemDataPos >= self->MemData.size()) { gst_app_src_end_of_stream(appsrc); return; } // "read" the data it wants, up to the remaining amount guint8 *data = &self->MemData[self->MemDataPos]; size = std::min(size, self->MemData.size() - self->MemDataPos); self->MemDataPos += size; GstBuffer *buffer = gst_buffer_new(); GST_BUFFER_DATA(buffer) = data; GST_BUFFER_SIZE(buffer) = size; // this takes ownership of the buffer; don't unref gst_app_src_push_buffer(appsrc, buffer); } static gboolean seek_memdata(GstAppSrc *appsrc, guint64 position, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); if(position > self->MemData.size()) return FALSE; self->MemDataPos = position; return TRUE; } static void memdata_source(GObject *object, GObject *orig, GParamSpec *pspec, OpenALSoundStream *self) { GstElement *elem; g_object_get(self->gstPipeline, "source", &elem, NULL); GstAppSrc *appsrc = GST_APP_SRC(elem); GstAppSrcCallbacks callbacks = { need_memdata, NULL, seek_memdata }; gst_app_src_set_callbacks(appsrc, &callbacks, self, NULL); gst_app_src_set_size(appsrc, self->MemData.size()); gst_app_src_set_stream_type(appsrc, GST_APP_STREAM_TYPE_RANDOM_ACCESS); gst_object_unref(appsrc); } // Callback-based data source SoundStreamCallback Callback; void *UserData; int BufferBytes; GstCaps *SrcCaps; static void need_callback(GstAppSrc *appsrc, guint size, gpointer user_data) { OpenALSoundStream *self = static_cast(user_data); GstBuffer *buffer; if(!self->Playing) buffer = gst_buffer_new_and_alloc(0); else { buffer = gst_buffer_new_and_alloc(size); if(!self->Callback(self, GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer), self->UserData)) { gst_buffer_unref(buffer); gst_app_src_end_of_stream(appsrc); return; } } gst_app_src_push_buffer(appsrc, buffer); } static void callback_source(GObject *object, GObject *orig, GParamSpec *pspec, OpenALSoundStream *self) { GstElement *elem; g_object_get(self->gstPipeline, "source", &elem, NULL); GstAppSrc *appsrc = GST_APP_SRC(elem); GstAppSrcCallbacks callbacks = { need_callback, NULL, NULL }; gst_app_src_set_callbacks(appsrc, &callbacks, self, NULL); gst_app_src_set_size(appsrc, -1); gst_app_src_set_max_bytes(appsrc, self->BufferBytes); gst_app_src_set_stream_type(appsrc, GST_APP_STREAM_TYPE_STREAM); gst_app_src_set_caps(appsrc, self->SrcCaps); gst_object_unref(appsrc); } // General methods virtual bool SetupSource() { // We don't actually use this source if we have an openalsink. However, // holding on to it helps ensure we don't overrun our allotted voice // count. if(Renderer->FreeSfx.size() == 0) { FSoundChan *lowest = Renderer->FindLowestChannel(); if(lowest) Renderer->StopChannel(lowest); if(Renderer->FreeSfx.size() == 0) return false; } Source = Renderer->FreeSfx.back(); Renderer->FreeSfx.pop_back(); alSource3f(Source, AL_DIRECTION, 0.f, 0.f, 0.f); alSource3f(Source, AL_VELOCITY, 0.f, 0.f, 0.f); alSource3f(Source, AL_POSITION, 0.f, 0.f, 0.f); alSourcef(Source, AL_MAX_GAIN, 1.f); alSourcef(Source, AL_GAIN, 1.f); alSourcef(Source, AL_PITCH, 1.f); alSourcef(Source, AL_ROLLOFF_FACTOR, 0.f); alSourcef(Source, AL_SEC_OFFSET, 0.f); alSourcei(Source, AL_SOURCE_RELATIVE, AL_TRUE); alSourcei(Source, AL_LOOPING, AL_FALSE); if(Renderer->EnvSlot) { alSourcef(Source, AL_ROOM_ROLLOFF_FACTOR, 0.f); alSourcef(Source, AL_AIR_ABSORPTION_FACTOR, 0.f); alSourcei(Source, AL_DIRECT_FILTER, AL_FILTER_NULL); alSource3i(Source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL); } return (getALError() == AL_NO_ERROR); } bool PipelineSetup() { TagList = gst_tag_list_new(); g_return_val_if_fail(TagList != NULL, false); // Flags (can be combined): // 0x01: video - Render the video stream // 0x02: audio - Render the audio stream // 0x04: text - Render subtitles // 0x08: vis - Render visualisation when no video is present // 0x10: soft-volume - Use software volume // 0x20: native-audio - Only use native audio formats // 0x40: native-video - Only use native video formats // 0x80: download - Attempt progressive download buffering int flags = 0x02 | 0x10; gstPipeline = gst_element_factory_make("playbin2", NULL); g_return_val_if_fail(gstPipeline != NULL, false); GstElement *sink = gst_element_factory_make("openalsink", NULL); if(sink != NULL) { // Give the sink our device, so it can create its own context and // source to play with (and not risk cross-contaminating errors) g_object_set(sink, "device-handle", Renderer->Device, NULL); } else { static bool warned = false; if(!warned) g_warning("Could not create an openalsink\n"); warned = true; sink = gst_element_factory_make("appsink", NULL); g_return_val_if_fail(sink != NULL, false); GstAppSink *appsink = GST_APP_SINK(sink); GstAppSinkCallbacks callbacks = { sink_eos, sink_preroll, sink_buffer, NULL }; GstCaps *caps = SupportedBufferFormatCaps(); gst_app_sink_set_callbacks(appsink, &callbacks, this, NULL); gst_app_sink_set_drop(appsink, FALSE); gst_app_sink_set_caps(appsink, caps); gst_caps_unref(caps); } // This takes ownership of the element; don't unref it g_object_set(gstPipeline, "audio-sink", sink, NULL); g_object_set(gstPipeline, "flags", flags, NULL); return true; } void HandleLoopTags() { // FIXME: Sample offsets assume a 44.1khz rate. Need to find some way // to get the actual rate of the file from GStreamer bool looppt_is_samples; unsigned int looppt; gchar *valstr; LoopPts[0] = 0; if(gst_tag_list_get_string(TagList, "LOOP_START", &valstr)) { g_print("Got LOOP_START string: %s\n", valstr); if(!S_ParseTimeTag(valstr, &looppt_is_samples, &looppt)) Printf("Invalid LOOP_START tag: '%s'\n", valstr); else LoopPts[0] = (looppt_is_samples ? ((gint64)looppt*1000000000/44100) : ((gint64)looppt*1000000)); g_free(valstr); } LoopPts[1] = -1; if(gst_tag_list_get_string(TagList, "LOOP_END", &valstr)) { g_print("Got LOOP_END string: %s\n", valstr); if(!S_ParseTimeTag(valstr, &looppt_is_samples, &looppt)) Printf("Invalid LOOP_END tag: '%s'\n", valstr); else { LoopPts[1] = (looppt_is_samples ? ((gint64)looppt*1000000000/44100) : ((gint64)looppt*1000000)); if(LoopPts[1] <= LoopPts[0]) LoopPts[1] = -1; } g_free(valstr); } } bool PreparePipeline() { GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return false; GstStateChangeReturn ret = gst_element_set_state(gstPipeline, GST_STATE_PAUSED); if(ret == GST_STATE_CHANGE_ASYNC) { const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_TAG | GST_MESSAGE_ASYNC_DONE; GstMessage *msg; while((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR) { PrintErrMsg("Prepare Error", msg); ret = GST_STATE_CHANGE_FAILURE; gst_message_unref(msg); break; } else if(GST_MESSAGE_TYPE(msg) != GST_MESSAGE_TAG) { gst_message_unref(msg); break; } GstTagList *tags = NULL; gst_message_parse_tag(msg, &tags); gst_tag_list_insert(TagList, tags, GST_TAG_MERGE_KEEP); gst_tag_list_free(tags); gst_message_unref(msg); } } else if(ret == GST_STATE_CHANGE_SUCCESS) { GstMessage *msg; while((msg=gst_bus_pop(bus)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_TAG) { GstTagList *tags = NULL; gst_message_parse_tag(msg, &tags); gst_tag_list_insert(TagList, tags, GST_TAG_MERGE_KEEP); gst_tag_list_free(tags); } gst_message_unref(msg); } } HandleLoopTags(); gst_object_unref(bus); bus = NULL; return (ret != GST_STATE_CHANGE_FAILURE); } public: FTempFileName tmpfile; ALfloat Volume; OpenALSoundStream(OpenALSoundRenderer *renderer) : Renderer(renderer), gstPipeline(NULL), TagList(NULL), Source(0), Playing(false), Looping(false), SamplesQueued(0), Callback(NULL), UserData(NULL), BufferBytes(0), SrcCaps(NULL), Volume(1.0f) { LoopPts[0] = LoopPts[1] = 0; Renderer->Streams.push_back(this); } virtual ~OpenALSoundStream() { Playing = false; if(SrcCaps) gst_caps_unref(SrcCaps); SrcCaps = NULL; if(gstPipeline) { gst_element_set_state(gstPipeline, GST_STATE_NULL); gst_object_unref(gstPipeline); gstPipeline = NULL; } if(TagList) gst_tag_list_free(TagList); TagList = NULL; if(Source) { alSourceRewind(Source); alSourcei(Source, AL_BUFFER, 0); Renderer->FreeSfx.push_back(Source); Source = 0; } if(Buffers.size() > 0) { alDeleteBuffers(Buffers.size(), &Buffers[0]); Buffers.clear(); } getALError(); Renderer->Streams.erase(find(Renderer->Streams.begin(), Renderer->Streams.end(), this)); Renderer = NULL; } virtual bool Play(bool looping, float vol) { if(Playing) return true; GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return false; Looping = looping; SetVolume(vol); if(Looping) { const GstSeekFlags flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_SEGMENT; gst_element_seek(gstPipeline, 1.0, GST_FORMAT_TIME, flags, GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_SET, LoopPts[1]); } // Start playing the stream Playing = true; GstStateChangeReturn ret = gst_element_set_state(gstPipeline, GST_STATE_PLAYING); if(ret == GST_STATE_CHANGE_FAILURE) Playing = false; if(ret == GST_STATE_CHANGE_ASYNC) { const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_ASYNC_DONE; GstMessage *msg; if((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR) { PrintErrMsg("Play Error", msg); Playing = false; } gst_message_unref(msg); } } gst_object_unref(bus); bus = NULL; return Playing; } virtual void Stop() { GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return; // Stop the stream GstStateChangeReturn ret = gst_element_set_state(gstPipeline, GST_STATE_PAUSED); if(ret == GST_STATE_CHANGE_ASYNC) { Playing = false; // Wait for the state change before requesting a seek const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_ASYNC_DONE; GstMessage *msg; if((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR) PrintErrMsg("Stop Error", msg); else if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ASYNC_DONE) ret = GST_STATE_CHANGE_SUCCESS; gst_message_unref(msg); } } if(ret == GST_STATE_CHANGE_SUCCESS) { Playing = false; alSourceRewind(Source); getALError(); const GstSeekFlags flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT; gst_element_seek_simple(gstPipeline, GST_FORMAT_TIME, flags, 0); } gst_object_unref(bus); bus = NULL; } virtual bool SetPaused(bool paused) { GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return false; GstStateChangeReturn ret; ret = gst_element_set_state(gstPipeline, (paused ? GST_STATE_PAUSED : GST_STATE_PLAYING)); if(ret == GST_STATE_CHANGE_ASYNC) { const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_ASYNC_DONE; GstMessage *msg; if((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR) { PrintErrMsg("Pause Error", msg); ret = GST_STATE_CHANGE_FAILURE; } gst_message_unref(msg); } } if(ret != GST_STATE_CHANGE_FAILURE && paused) { alSourcePause(Source); getALError(); } gst_object_unref(bus); bus = NULL; return (ret != GST_STATE_CHANGE_FAILURE); } virtual void SetVolume(float vol) { Volume = vol; g_object_set(gstPipeline, "volume", (double)(Volume*Renderer->MusicVolume), NULL); } virtual unsigned int GetPosition() { GstFormat format = GST_FORMAT_TIME; gint64 pos; // Position will be handled in milliseconds; GStreamer's time format is in nanoseconds if(gst_element_query_position(gstPipeline, &format, &pos) && format == GST_FORMAT_TIME) return (unsigned int)(pos / 1000000); return 0; } virtual bool SetPosition(unsigned int val) { gint64 pos = (gint64)val * 1000000; return gst_element_seek_simple(gstPipeline, GST_FORMAT_TIME, GST_SEEK_FLAG_ACCURATE, pos); } virtual bool IsEnded() { GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return true; GstMessage *msg; while((msg=gst_bus_pop(bus)) != NULL) { switch(GST_MESSAGE_TYPE(msg)) { case GST_MESSAGE_SEGMENT_DONE: case GST_MESSAGE_EOS: Playing = false; if(Looping) Playing = gst_element_seek(gstPipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_SEGMENT, GST_SEEK_TYPE_SET, LoopPts[0], GST_SEEK_TYPE_SET, LoopPts[1]); break; case GST_MESSAGE_ERROR: PrintErrMsg("Pipeline Error", msg); Playing = false; break; case GST_MESSAGE_WARNING: PrintErrMsg("Pipeline Warning", msg); break; default: break; } gst_message_unref(msg); } gst_object_unref(bus); bus = NULL; return !Playing; } bool Init(const char *filename) { if(!SetupSource() || !PipelineSetup()) return false; GError *err = NULL; gchar *uri; if(g_path_is_absolute(filename)) uri = g_filename_to_uri(filename, NULL, &err); else if(g_strrstr(filename, "://") != NULL) uri = g_strdup(filename); else { gchar *curdir = g_get_current_dir(); gchar *absolute_path = g_strconcat(curdir, G_DIR_SEPARATOR_S, filename, NULL); uri = g_filename_to_uri(absolute_path, NULL, &err); g_free(absolute_path); g_free(curdir); } if(!uri) { if(err) { Printf("Failed to convert "TEXTCOLOR_BOLD"%s"TEXTCOLOR_NORMAL" to URI: %s\n", filename, err->message); g_error_free(err); } return false; } g_object_set(gstPipeline, "uri", uri, NULL); g_free(uri); return PreparePipeline(); } bool Init(const BYTE *data, unsigned int datalen) { // Can't keep the original pointer since the memory can apparently be // overwritten at some point (happens with MIDI data, at least) MemData.resize(datalen); memcpy(&MemData[0], data, datalen); MemDataPos = 0; if(!SetupSource() || !PipelineSetup()) return false; g_object_set(gstPipeline, "uri", "appsrc://", NULL); g_signal_connect(gstPipeline, "deep-notify::source", G_CALLBACK(memdata_source), this); return PreparePipeline(); } bool Init(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata) { Callback = callback; UserData = userdata; BufferBytes = buffbytes; GstStructure *structure; if((flags&Float)) structure = gst_structure_new("audio/x-raw-float", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); else if((flags&Bits32)) structure = gst_structure_new("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, "depth", G_TYPE_INT, 32, "signed", G_TYPE_BOOLEAN, TRUE, NULL); else if((flags&Bits8)) structure = gst_structure_new("audio/x-raw-int", "width", G_TYPE_INT, 8, "depth", G_TYPE_INT, 8, "signed", G_TYPE_BOOLEAN, TRUE, NULL); else structure = gst_structure_new("audio/x-raw-int", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, "signed", G_TYPE_BOOLEAN, TRUE, NULL); gst_structure_set(structure, "channels", G_TYPE_INT, (flags&Mono)?1:2, NULL); gst_structure_set(structure, "rate", G_TYPE_INT, samplerate, NULL); SrcCaps = gst_caps_new_full(structure, NULL); if(!SrcCaps || !SetupSource() || !PipelineSetup()) return false; g_object_set(gstPipeline, "uri", "appsrc://", NULL); g_signal_connect(gstPipeline, "deep-notify::source", G_CALLBACK(callback_source), this); return PreparePipeline(); } }; class Decoder { GstElement *gstPipeline, *gstSink; GstTagList *TagList; const guint8 *MemData; size_t MemDataSize; size_t MemDataPos; static void need_memdata(GstAppSrc *appsrc, guint size, gpointer user_data) { Decoder *self = static_cast(user_data); GstFlowReturn ret; if(self->MemDataPos >= self->MemDataSize) { gst_app_src_end_of_stream(appsrc); return; } const guint8 *data = &self->MemData[self->MemDataPos]; size = (std::min)(size, (guint)(self->MemDataSize - self->MemDataPos)); self->MemDataPos += size; GstBuffer *buffer = gst_buffer_new(); GST_BUFFER_DATA(buffer) = const_cast(data); GST_BUFFER_SIZE(buffer) = size; gst_app_src_push_buffer(appsrc, buffer); } static gboolean seek_memdata(GstAppSrc *appsrc, guint64 position, gpointer user_data) { Decoder *self = static_cast(user_data); if(position > self->MemDataSize) return FALSE; self->MemDataPos = position; return TRUE; } static void memdata_source(GObject *object, GObject *orig, GParamSpec *pspec, Decoder *self) { GstElement *elem; g_object_get(self->gstPipeline, "source", &elem, NULL); GstAppSrc *appsrc = GST_APP_SRC(elem); GstAppSrcCallbacks callbacks = { need_memdata, NULL, seek_memdata }; gst_app_src_set_callbacks(appsrc, &callbacks, self, NULL); gst_app_src_set_size(appsrc, self->MemDataSize); gst_app_src_set_stream_type(appsrc, GST_APP_STREAM_TYPE_RANDOM_ACCESS); gst_object_unref(appsrc); } static GstFlowReturn sink_preroll(GstAppSink *sink, gpointer user_data) { Decoder *self = static_cast(user_data); GstBuffer *buffer = gst_app_sink_pull_preroll(sink); if(!buffer) return GST_FLOW_ERROR; if(self->OutRate == 0) { GstCaps *caps = GST_BUFFER_CAPS(buffer); gint channels = 0, rate = 0, bits = 0, i; for(i = gst_caps_get_size(caps)-1;i >= 0;i--) { GstStructure *struc = gst_caps_get_structure(caps, i); if(gst_structure_has_field(struc, "channels")) gst_structure_get_int(struc, "channels", &channels); if(gst_structure_has_field(struc, "rate")) gst_structure_get_int(struc, "rate", &rate); if(gst_structure_has_field(struc, "width")) gst_structure_get_int(struc, "width", &bits); } self->OutChannels = channels; self->OutBits = bits; self->OutRate = rate; } gst_buffer_unref(buffer); if(self->OutRate <= 0) return GST_FLOW_ERROR; return GST_FLOW_OK; } static GstFlowReturn sink_buffer(GstAppSink *sink, gpointer user_data) { Decoder *self = static_cast(user_data); GstBuffer *buffer = gst_app_sink_pull_buffer(sink); if(!buffer) return GST_FLOW_ERROR; guint newsize = GST_BUFFER_SIZE(buffer); size_t pos = self->OutData.size(); self->OutData.resize(pos+newsize); memcpy(&self->OutData[pos], GST_BUFFER_DATA(buffer), newsize); gst_buffer_unref(buffer); return GST_FLOW_OK; } bool PipelineSetup(int forcebits) { if(forcebits && forcebits != 8 && forcebits != 16) return false; TagList = gst_tag_list_new(); g_return_val_if_fail(TagList != NULL, false); gstPipeline = gst_element_factory_make("playbin2", NULL); g_return_val_if_fail(gstPipeline != NULL, false); gstSink = gst_element_factory_make("appsink", NULL); g_return_val_if_fail(gstSink != NULL, false); GstAppSink *appsink = GST_APP_SINK(gstSink); GstAppSinkCallbacks callbacks = { NULL, sink_preroll, sink_buffer, NULL }; GstCaps *caps = SupportedBufferFormatCaps(forcebits); g_object_set(appsink, "sync", FALSE, NULL); gst_app_sink_set_callbacks(appsink, &callbacks, this, NULL); gst_app_sink_set_drop(appsink, FALSE); gst_app_sink_set_caps(appsink, caps); g_object_set(gstPipeline, "audio-sink", gst_object_ref(gstSink), NULL); g_object_set(gstPipeline, "flags", 0x02, NULL); gst_caps_unref(caps); return true; } void HandleLoopTags(unsigned int looppt[2], bool looppt_is_samples[2], bool has_looppt[2]) { gchar *valstr; if(gst_tag_list_get_string(TagList, "LOOP_START", &valstr)) { g_print("Got LOOP_START string: %s\n", valstr); has_looppt[0] = S_ParseTimeTag(valstr, &looppt_is_samples[0], &looppt[0]); if(!has_looppt[0]) Printf("Invalid LOOP_START tag: '%s'\n", valstr); g_free(valstr); } if(gst_tag_list_get_string(TagList, "LOOP_END", &valstr)) { g_print("Got LOOP_END string: %s\n", valstr); has_looppt[1] = S_ParseTimeTag(valstr, &looppt_is_samples[1], &looppt[1]); if(!has_looppt[1]) Printf("Invalid LOOP_END tag: '%s'\n", valstr); g_free(valstr); } } public: std::vector OutData; ALint LoopPts[2]; ALsizei OutRate; ALuint OutChannels; ALuint OutBits; Decoder() : gstPipeline(NULL), gstSink(NULL), TagList(NULL), OutRate(0), OutChannels(0), OutBits(0) { LoopPts[0] = LoopPts[1] = 0; } virtual ~Decoder() { if(gstSink) gst_object_unref(gstSink); gstSink = NULL; if(gstPipeline) { gst_element_set_state(gstPipeline, GST_STATE_NULL); gst_object_unref(gstPipeline); gstPipeline = NULL; } if(TagList) gst_tag_list_free(TagList); TagList = NULL; } bool Decode(const void *data, unsigned int datalen, int forcebits=0) { MemData = static_cast(data); MemDataSize = datalen; MemDataPos = 0; OutData.clear(); if(!PipelineSetup(forcebits)) return false; g_object_set(gstPipeline, "uri", "appsrc://", NULL); g_signal_connect(gstPipeline, "deep-notify::source", G_CALLBACK(memdata_source), this); GstBus *bus = gst_element_get_bus(gstPipeline); if(!bus) return false; GstMessage *msg; GstStateChangeReturn ret = gst_element_set_state(gstPipeline, GST_STATE_PLAYING); if(ret == GST_STATE_CHANGE_ASYNC) { const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_TAG | GST_MESSAGE_ASYNC_DONE; while((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ASYNC_DONE) { ret = GST_STATE_CHANGE_SUCCESS; break; } else if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_TAG) { GstTagList *tags = NULL; gst_message_parse_tag(msg, &tags); gst_tag_list_insert(TagList, tags, GST_TAG_MERGE_KEEP); gst_tag_list_free(tags); } else if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR) { ret = GST_STATE_CHANGE_FAILURE; PrintErrMsg("Decoder Error", msg); break; } gst_message_unref(msg); } } bool err = true; if(ret == GST_STATE_CHANGE_SUCCESS) { const GstMessageType types = GST_MESSAGE_ERROR | GST_MESSAGE_TAG | GST_MESSAGE_EOS; while((msg=gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, types)) != NULL) { if(GST_MESSAGE_TYPE(msg) == GST_MESSAGE_EOS) { err = false; gst_message_unref(msg); break; } if(GST_MESSAGE_TYPE(msg) != GST_MESSAGE_TAG) { PrintErrMsg("Decoder Error", msg); gst_message_unref(msg); break; } GstTagList *tags = NULL; gst_message_parse_tag(msg, &tags); gst_tag_list_insert(TagList, tags, GST_TAG_MERGE_KEEP); gst_tag_list_free(tags); gst_message_unref(msg); } } if(!err) { ALuint FrameSize = OutChannels*OutBits/8; if(OutData.size() >= FrameSize) { // HACK: Evilness abound. Seems GStreamer doesn't like (certain?) // wave files and produces an extra sample, which can cause an // audible click at the end. Cut it. OutData.resize(OutData.size() - FrameSize); } unsigned int looppt[2] = { 0, 0 }; bool looppt_is_samples[2] = { true, true }; bool has_looppt[2] = { false, false }; HandleLoopTags(looppt, looppt_is_samples, has_looppt); if(has_looppt[0] || has_looppt[1]) { if(!has_looppt[0]) LoopPts[0] = 0; else if(looppt_is_samples[0]) LoopPts[0] = (std::min)((ALint)looppt[0], (ALint)(OutData.size() / FrameSize)); else LoopPts[0] = (std::min)((ALint)((gint64)looppt[0] * OutRate / 1000), (ALint)(OutData.size() / FrameSize)); if(!has_looppt[1]) LoopPts[1] = OutData.size() / FrameSize; else if(looppt_is_samples[1]) LoopPts[1] = (std::min)((ALint)looppt[1], (ALint)(OutData.size() / FrameSize)); else LoopPts[1] = (std::min)((ALint)((gint64)looppt[1] * OutRate / 1000), (ALint)(OutData.size() / FrameSize)); } } gst_object_unref(bus); bus = NULL; return !err; } }; #else /* WITH_GSTREAMER */ class OpenALSoundStream : public SoundStream { OpenALSoundRenderer *Renderer; public: FTempFileName tmpfile; ALfloat Volume; OpenALSoundStream(OpenALSoundRenderer *renderer) : Renderer(renderer), Volume(1.0f) { Renderer->Streams.push_back(this); } virtual ~OpenALSoundStream() { Renderer->Streams.erase(find(Renderer->Streams.begin(), Renderer->Streams.end(), this)); Renderer = NULL; } virtual bool Play(bool, float) { return false; } virtual void Stop() { } virtual void SetVolume(float vol) { Volume = vol; } virtual bool SetPaused(bool) { return false; } virtual unsigned int GetPosition() { return 0; } virtual bool IsEnded() { return true; } bool Init(const char*) { return false; } bool Init(const BYTE*, unsigned int) { return false; } bool Init(SoundStreamCallback, int, int, int, void*) { return false; } }; class Decoder { public: std::vector OutData; ALint LoopPts[2]; ALsizei OutRate; ALuint OutChannels; ALuint OutBits; Decoder() : OutRate(0), OutChannels(0), OutBits(0) { LoopPts[0] = LoopPts[1] = 0; } bool Decode(const void*, unsigned int, int=0) { return false; } }; #endif /* WITH_GSTREAMER */ template static void LoadALFunc(const char *name, T *x) { *x = reinterpret_cast(alGetProcAddress(name)); } OpenALSoundRenderer::OpenALSoundRenderer() : Device(NULL), Context(NULL), SrcDistanceModel(false), SFXPaused(0), PrevEnvironment(NULL), EnvSlot(0) { EnvFilters[0] = EnvFilters[1] = 0; Printf("I_InitSound: Initializing OpenAL\n"); #ifdef WITH_GSTREAMER static bool GSTInited = false; if(!GSTInited) { GError *error; if(!gst_init_check(NULL, NULL, &error)) { Printf("Failed to initialize GStreamer: %s\n", error->message); g_error_free(error); return; } GSTInited = true; } #endif if(snd_aldevice != "Default") { Device = alcOpenDevice(*snd_aldevice); if(!Device) Printf(TEXTCOLOR_BLUE" Failed to open device "TEXTCOLOR_BOLD"%s"TEXTCOLOR_BLUE". Trying default.\n", *snd_aldevice); } if(!Device) { Device = alcOpenDevice(NULL); if(!Device) { Printf(TEXTCOLOR_RED" Could not open audio device\n"); return; } } Printf(" Opened device "TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_DEVICE_SPECIFIER)); ALCint major=0, minor=0; alcGetIntegerv(Device, ALC_MAJOR_VERSION, 1, &major); alcGetIntegerv(Device, ALC_MINOR_VERSION, 1, &minor); DPrintf(" ALC Version: "TEXTCOLOR_BLUE"%d.%d\n", major, minor); DPrintf(" ALC Extensions: "TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_EXTENSIONS)); DisconnectNotify = alcIsExtensionPresent(Device, "ALC_EXT_disconnect"); std::vector attribs; if(*snd_samplerate > 0) { attribs.push_back(ALC_FREQUENCY); attribs.push_back(*snd_samplerate); } // Make sure one source is capable of stereo output with the rest doing // mono, without running out of voices attribs.push_back(ALC_MONO_SOURCES); attribs.push_back((std::max)(*snd_channels, 2) - 1); attribs.push_back(ALC_STEREO_SOURCES); attribs.push_back(1); // Other attribs..? attribs.push_back(0); Context = alcCreateContext(Device, &attribs[0]); if(!Context || alcMakeContextCurrent(Context) == ALC_FALSE) { Printf(TEXTCOLOR_RED" Failed to setup context: %s\n", alcGetString(Device, alcGetError(Device))); if(Context) alcDestroyContext(Context); Context = NULL; alcCloseDevice(Device); Device = NULL; return; } attribs.clear(); DPrintf(" Vendor: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VENDOR)); DPrintf(" Renderer: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_RENDERER)); DPrintf(" Version: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VERSION)); DPrintf(" Extensions: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_EXTENSIONS)); SrcDistanceModel = alIsExtensionPresent("AL_EXT_source_distance_model"); LoopPoints = alIsExtensionPresent("AL_EXT_loop_points"); alDopplerFactor(0.5f); alSpeedOfSound(343.3f * 96.0f); alDistanceModel(AL_INVERSE_DISTANCE); if(SrcDistanceModel) alEnable(AL_SOURCE_DISTANCE_MODEL); ALenum err = getALError(); if(err != AL_NO_ERROR) { alcMakeContextCurrent(NULL); alcDestroyContext(Context); Context = NULL; alcCloseDevice(Device); Device = NULL; return; } ALCint numMono=0, numStereo=0; alcGetIntegerv(Device, ALC_MONO_SOURCES, 1, &numMono); alcGetIntegerv(Device, ALC_STEREO_SOURCES, 1, &numStereo); Sources.resize((std::min)((std::max)(*snd_channels, 2), numMono+numStereo)); for(size_t i = 0;i < Sources.size();i++) { alGenSources(1, &Sources[i]); if(getALError() != AL_NO_ERROR) { Sources.resize(i); break; } FreeSfx.push_back(Sources[i]); } if(Sources.size() == 0) { Printf(TEXTCOLOR_RED" Error: could not generate any sound sources!\n"); alcMakeContextCurrent(NULL); alcDestroyContext(Context); Context = NULL; alcCloseDevice(Device); Device = NULL; return; } DPrintf(" Allocated "TEXTCOLOR_BLUE"%u"TEXTCOLOR_NORMAL" sources\n", Sources.size()); LastWaterAbsorb = 0.0f; if(*snd_efx && alcIsExtensionPresent(Device, "ALC_EXT_EFX")) { // EFX function pointers #define LOAD_FUNC(x) (LoadALFunc(#x, &x)) LOAD_FUNC(alGenEffects); LOAD_FUNC(alDeleteEffects); LOAD_FUNC(alIsEffect); LOAD_FUNC(alEffecti); LOAD_FUNC(alEffectiv); LOAD_FUNC(alEffectf); LOAD_FUNC(alEffectfv); LOAD_FUNC(alGetEffecti); LOAD_FUNC(alGetEffectiv); LOAD_FUNC(alGetEffectf); LOAD_FUNC(alGetEffectfv); LOAD_FUNC(alGenFilters); LOAD_FUNC(alDeleteFilters); LOAD_FUNC(alIsFilter); LOAD_FUNC(alFilteri); LOAD_FUNC(alFilteriv); LOAD_FUNC(alFilterf); LOAD_FUNC(alFilterfv); LOAD_FUNC(alGetFilteri); LOAD_FUNC(alGetFilteriv); LOAD_FUNC(alGetFilterf); LOAD_FUNC(alGetFilterfv); LOAD_FUNC(alGenAuxiliaryEffectSlots); LOAD_FUNC(alDeleteAuxiliaryEffectSlots); LOAD_FUNC(alIsAuxiliaryEffectSlot); LOAD_FUNC(alAuxiliaryEffectSloti); LOAD_FUNC(alAuxiliaryEffectSlotiv); LOAD_FUNC(alAuxiliaryEffectSlotf); LOAD_FUNC(alAuxiliaryEffectSlotfv); LOAD_FUNC(alGetAuxiliaryEffectSloti); LOAD_FUNC(alGetAuxiliaryEffectSlotiv); LOAD_FUNC(alGetAuxiliaryEffectSlotf); LOAD_FUNC(alGetAuxiliaryEffectSlotfv); #undef LOAD_FUNC if(getALError() == AL_NO_ERROR) { ALuint envReverb; alGenEffects(1, &envReverb); if(getALError() == AL_NO_ERROR) { alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB); if(alGetError() == AL_NO_ERROR) DPrintf(" EAX Reverb found\n"); alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB); if(alGetError() == AL_NO_ERROR) DPrintf(" Standard Reverb found\n"); alDeleteEffects(1, &envReverb); getALError(); } alGenAuxiliaryEffectSlots(1, &EnvSlot); alGenFilters(2, EnvFilters); if(getALError() == AL_NO_ERROR) { alFilteri(EnvFilters[0], AL_FILTER_TYPE, AL_FILTER_LOWPASS); alFilteri(EnvFilters[1], AL_FILTER_TYPE, AL_FILTER_LOWPASS); if(getALError() == AL_NO_ERROR) DPrintf(" Lowpass found\n"); else { alDeleteFilters(2, EnvFilters); EnvFilters[0] = EnvFilters[1] = 0; alDeleteAuxiliaryEffectSlots(1, &EnvSlot); EnvSlot = 0; getALError(); } } else { alDeleteFilters(2, EnvFilters); alDeleteAuxiliaryEffectSlots(1, &EnvSlot); EnvFilters[0] = EnvFilters[1] = 0; EnvSlot = 0; getALError(); } } } if(EnvSlot) Printf(" EFX enabled\n"); snd_sfxvolume.Callback(); } OpenALSoundRenderer::~OpenALSoundRenderer() { if(!Device) return; while(Streams.size() > 0) delete Streams[0]; alDeleteSources(Sources.size(), &Sources[0]); Sources.clear(); FreeSfx.clear(); SfxGroup.clear(); PausableSfx.clear(); ReverbSfx.clear(); for(EffectMap::iterator i = EnvEffects.begin();i != EnvEffects.end();i++) { if(i->second) alDeleteEffects(1, &(i->second)); } EnvEffects.clear(); if(EnvSlot) { alDeleteAuxiliaryEffectSlots(1, &EnvSlot); alDeleteFilters(2, EnvFilters); } EnvSlot = 0; EnvFilters[0] = EnvFilters[1] = 0; alcMakeContextCurrent(NULL); alcDestroyContext(Context); Context = NULL; alcCloseDevice(Device); Device = NULL; } void OpenALSoundRenderer::SetSfxVolume(float volume) { SfxVolume = volume; FSoundChan *schan = Channels; while(schan) { if(schan->SysChannel != NULL) { ALuint source = *((ALuint*)schan->SysChannel); volume = SfxVolume; alcSuspendContext(Context); alSourcef(source, AL_MAX_GAIN, volume); if(schan->ManualGain) volume *= GetRolloff(&schan->Rolloff, sqrt(schan->DistanceSqr)); alSourcef(source, AL_GAIN, volume); } schan = schan->NextChan; } getALError(); } void OpenALSoundRenderer::SetMusicVolume(float volume) { MusicVolume = volume; foreach(OpenALSoundStream*, i, Streams) (*i)->SetVolume((*i)->Volume); } unsigned int OpenALSoundRenderer::GetMSLength(SoundHandle sfx) { if(sfx.data) { ALuint buffer = *((ALuint*)sfx.data); if(alIsBuffer(buffer)) { ALint bits, channels, freq, size; alGetBufferi(buffer, AL_BITS, &bits); alGetBufferi(buffer, AL_CHANNELS, &channels); alGetBufferi(buffer, AL_FREQUENCY, &freq); alGetBufferi(buffer, AL_SIZE, &size); if(getALError() == AL_NO_ERROR) return (unsigned int)(size / (channels*bits/8) * 1000. / freq); } } return 0; } unsigned int OpenALSoundRenderer::GetSampleLength(SoundHandle sfx) { if(!sfx.data) return 0; return getBufferLength(*((ALuint*)sfx.data)); } float OpenALSoundRenderer::GetOutputRate() { ALCint rate = 44100; // Default, just in case alcGetIntegerv(Device, ALC_FREQUENCY, 1, &rate); return (float)rate; } SoundHandle OpenALSoundRenderer::LoadSoundRaw(BYTE *sfxdata, int length, int frequency, int channels, int bits, int loopstart) { SoundHandle retval = { NULL }; if(length == 0) return retval; if(bits == -8) { // Simple signed->unsigned conversion for(int i = 0;i < length;i++) sfxdata[i] ^= 0x80; bits = -bits; } ALenum format = AL_NONE; if(bits == 16) { if(channels == 1) format = AL_FORMAT_MONO16; if(channels == 2) format = AL_FORMAT_STEREO16; } else if(bits == 8) { if(channels == 1) format = AL_FORMAT_MONO8; if(channels == 2) format = AL_FORMAT_STEREO8; } if(format == AL_NONE || frequency <= 0) { Printf("Unhandled format: %d bit, %d channel, %d hz\n", bits, channels, frequency); return retval; } length -= length%(channels*bits/8); ALenum err; ALuint buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, sfxdata, length, frequency); if((err=getALError()) != AL_NO_ERROR) { Printf("Failed to buffer data: %s\n", alGetString(err)); alDeleteBuffers(1, &buffer); getALError(); return retval; } if(loopstart > 0 && LoopPoints) { ALint loops[2] = { loopstart, length / (channels*bits/8) }; Printf("Setting loop points %d -> %d\n", loops[0], loops[1]); alBufferiv(buffer, AL_LOOP_POINTS, loops); getALError(); } else if(loopstart > 0) { static bool warned = false; if(!warned) Printf("Loop points not supported!\n"); warned = true; } retval.data = new ALuint(buffer); return retval; } SoundHandle OpenALSoundRenderer::LoadSound(BYTE *sfxdata, int length) { SoundHandle retval = { NULL }; Decoder decoder; if(!decoder.Decode(sfxdata, length)) { DPrintf("Failed to decode sound\n"); return retval; } ALenum format = FormatFromDesc(decoder.OutBits, decoder.OutChannels); if(format == AL_NONE) { Printf("Unhandled format: %d bit, %d channel\n", decoder.OutBits, decoder.OutChannels); return retval; } ALenum err; ALuint buffer = 0; alGenBuffers(1, &buffer); alBufferData(buffer, format, &decoder.OutData[0], decoder.OutData.size(), decoder.OutRate); if((err=getALError()) != AL_NO_ERROR) { Printf("Failed to buffer data: %s\n", alGetString(err)); alDeleteBuffers(1, &buffer); getALError(); return retval; } if(LoopPoints && decoder.LoopPts[1] > decoder.LoopPts[0]) { alBufferiv(buffer, AL_LOOP_POINTS, decoder.LoopPts); getALError(); } else if(decoder.LoopPts[1] > decoder.LoopPts[0]) { static bool warned = false; if(!warned) Printf("Loop points not supported!\n"); warned = true; } retval.data = new ALuint(buffer); return retval; } void OpenALSoundRenderer::UnloadSound(SoundHandle sfx) { if(!sfx.data) return; FSoundChan *schan = Channels; while(schan) { if(schan->SysChannel) { ALint bufID = 0; alGetSourcei(*((ALuint*)schan->SysChannel), AL_BUFFER, &bufID); if(bufID == *((ALint*)sfx.data)) { FSoundChan *next = schan->NextChan; StopChannel(schan); schan = next; continue; } } schan = schan->NextChan; } alDeleteBuffers(1, ((ALuint*)sfx.data)); getALError(); delete ((ALuint*)sfx.data); } short *OpenALSoundRenderer::DecodeSample(int outlen, const void *coded, int sizebytes, ECodecType type) { Decoder decoder; // Force 16-bit if(!decoder.Decode(coded, sizebytes, 16)) { DPrintf("Failed to decode sample\n"); return NULL; } if(decoder.OutChannels != 1) { DPrintf("Sample is not mono\n"); return NULL; } short *samples = (short*)malloc(outlen); if((size_t)outlen > decoder.OutData.size()) { memcpy(samples, &decoder.OutData[0], decoder.OutData.size()); memset(&samples[decoder.OutData.size()/sizeof(short)], 0, outlen-decoder.OutData.size()); } else memcpy(samples, &decoder.OutData[0], outlen); return samples; } SoundStream *OpenALSoundRenderer::CreateStream(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata) { OpenALSoundStream *stream = new OpenALSoundStream(this); if(!stream->Init(callback, buffbytes, flags, samplerate, userdata)) { delete stream; stream = NULL; } return stream; } SoundStream *OpenALSoundRenderer::OpenStream(const char *filename, int flags, int offset, int length) { std::auto_ptr stream(new OpenALSoundStream(this)); if(offset > 0) { // If there's an offset to the start of the data, separate it into its // own temp file FILE *infile = fopen(filename, "rb"); FILE *f = fopen(stream->tmpfile, "wb"); if(!infile || !f || fseek(infile, offset, SEEK_SET) != 0) { if(infile) fclose(infile); if(f) fclose(f); return NULL; } BYTE buf[1024]; size_t got; do { got = (std::min)(sizeof(buf), (size_t)length); got = fread(buf, 1, got, infile); if(got == 0) break; } while(fwrite(buf, 1, got, f) == got && (length-=got) > 0); fclose(f); fclose(infile); filename = stream->tmpfile; } bool ok = ((offset == -1) ? stream->Init((const BYTE*)filename, length) : stream->Init(filename)); if(ok == false) return NULL; return stream.release(); } FISoundChannel *OpenALSoundRenderer::StartSound(SoundHandle sfx, float vol, int pitch, int chanflags, FISoundChannel *reuse_chan) { if(FreeSfx.size() == 0) { FSoundChan *lowest = FindLowestChannel(); if(lowest) StopChannel(lowest); if(FreeSfx.size() == 0) return NULL; } ALuint buffer = *((ALuint*)sfx.data); ALuint &source = *find(Sources.begin(), Sources.end(), FreeSfx.back()); alSource3f(source, AL_POSITION, 0.f, 0.f, 0.f); alSource3f(source, AL_VELOCITY, 0.f, 0.f, 0.f); alSource3f(source, AL_DIRECTION, 0.f, 0.f, 0.f); alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE); alSourcei(source, AL_LOOPING, (chanflags&SNDF_LOOP) ? AL_TRUE : AL_FALSE); alSourcef(source, AL_REFERENCE_DISTANCE, 1.f); alSourcef(source, AL_MAX_DISTANCE, 1000.f); alSourcef(source, AL_ROLLOFF_FACTOR, 0.f); alSourcef(source, AL_MAX_GAIN, SfxVolume); alSourcef(source, AL_GAIN, SfxVolume*vol); if(EnvSlot) { if(!(chanflags&SNDF_NOREVERB)) { alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]); alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]); alSourcef(source, AL_AIR_ABSORPTION_FACTOR, LastWaterAbsorb); } else { alSourcei(source, AL_DIRECT_FILTER, AL_FILTER_NULL); alSource3i(source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL); alSourcef(source, AL_AIR_ABSORPTION_FACTOR, 0.f); } alSourcef(source, AL_ROOM_ROLLOFF_FACTOR, 0.f); alSourcef(source, AL_PITCH, PITCH(pitch)); } else if(LastWaterAbsorb > 0.f && !(chanflags&SNDF_NOREVERB)) alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT); else alSourcef(source, AL_PITCH, PITCH(pitch)); if(!reuse_chan) alSourcef(source, AL_SEC_OFFSET, 0.f); else { if((chanflags&SNDF_ABSTIME)) alSourcef(source, AL_SEC_OFFSET, reuse_chan->StartTime.Lo/1000.f); else { // FIXME: set offset based on the current time and the StartTime alSourcef(source, AL_SEC_OFFSET, 0.f); } } if(getALError() != AL_NO_ERROR) return NULL; alSourcei(source, AL_BUFFER, buffer); if((chanflags&SNDF_NOPAUSE) || !SFXPaused) alSourcePlay(source); if(getALError() != AL_NO_ERROR) { alSourcei(source, AL_BUFFER, 0); getALError(); return NULL; } if(!(chanflags&SNDF_NOREVERB)) ReverbSfx.push_back(source); if(!(chanflags&SNDF_NOPAUSE)) PausableSfx.push_back(source); SfxGroup.push_back(source); FreeSfx.pop_back(); FISoundChannel *chan = reuse_chan; if(!chan) chan = S_GetChannel(&source); else chan->SysChannel = &source; chan->Rolloff.RolloffType = ROLLOFF_Linear; chan->Rolloff.MaxDistance = 2.f; chan->Rolloff.MinDistance = 1.f; chan->DistanceScale = 1.f; chan->DistanceSqr = (2.f-vol)*(2.f-vol); chan->ManualGain = true; return chan; } FISoundChannel *OpenALSoundRenderer::StartSound3D(SoundHandle sfx, SoundListener *listener, float vol, FRolloffInfo *rolloff, float distscale, int pitch, int priority, const FVector3 &pos, const FVector3 &vel, int channum, int chanflags, FISoundChannel *reuse_chan) { float dist_sqr = (pos - listener->position).LengthSquared() * distscale*distscale; if(FreeSfx.size() == 0) { FSoundChan *lowest = FindLowestChannel(); if(lowest) { if(lowest->Priority < priority || (lowest->Priority == priority && lowest->DistanceSqr > dist_sqr)) StopChannel(lowest); } if(FreeSfx.size() == 0) return NULL; } float rolloffFactor, gain; bool manualGain = true; ALuint buffer = *((ALuint*)sfx.data); ALint channels = 1; alGetBufferi(buffer, AL_CHANNELS, &channels); ALuint &source = *find(Sources.begin(), Sources.end(), FreeSfx.back()); alSource3f(source, AL_POSITION, pos[0], pos[1], -pos[2]); alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]); alSource3f(source, AL_DIRECTION, 0.f, 0.f, 0.f); alSourcei(source, AL_SOURCE_RELATIVE, AL_FALSE); alSourcei(source, AL_LOOPING, (chanflags&SNDF_LOOP) ? AL_TRUE : AL_FALSE); // Multi-channel sources won't attenuate in OpenAL, and "area sounds" have // special rolloff properties (they have a panning radius of 32 units, but // start attenuating at MinDistance). if(channels == 1 && !(chanflags&SNDF_AREA)) { if(rolloff->RolloffType == ROLLOFF_Log) { if(SrcDistanceModel) alSourcei(source, AL_DISTANCE_MODEL, AL_INVERSE_DISTANCE); alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale); alSourcef(source, AL_MAX_DISTANCE, (1000.f+rolloff->MinDistance)/distscale); rolloffFactor = rolloff->RolloffFactor; manualGain = false; gain = 1.f; } else if(rolloff->RolloffType == ROLLOFF_Linear && SrcDistanceModel) { alSourcei(source, AL_DISTANCE_MODEL, AL_LINEAR_DISTANCE); alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale); alSourcef(source, AL_MAX_DISTANCE, rolloff->MaxDistance/distscale); rolloffFactor = 1.f; manualGain = false; gain = 1.f; } } if(manualGain) { if(SrcDistanceModel) alSourcei(source, AL_DISTANCE_MODEL, AL_NONE); if((chanflags&SNDF_AREA) && rolloff->MinDistance < 32.f) alSourcef(source, AL_REFERENCE_DISTANCE, 32.f/distscale); else alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale); alSourcef(source, AL_MAX_DISTANCE, (1000.f+rolloff->MinDistance)/distscale); rolloffFactor = 0.f; gain = GetRolloff(rolloff, sqrt(dist_sqr)); } alSourcef(source, AL_ROLLOFF_FACTOR, rolloffFactor); alSourcef(source, AL_MAX_GAIN, SfxVolume); alSourcef(source, AL_GAIN, SfxVolume * gain); if(EnvSlot) { if(!(chanflags&SNDF_NOREVERB)) { alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]); alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]); alSourcef(source, AL_AIR_ABSORPTION_FACTOR, LastWaterAbsorb); } else { alSourcei(source, AL_DIRECT_FILTER, AL_FILTER_NULL); alSource3i(source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL); alSourcef(source, AL_AIR_ABSORPTION_FACTOR, 0.f); } alSourcef(source, AL_ROOM_ROLLOFF_FACTOR, rolloffFactor); alSourcef(source, AL_PITCH, PITCH(pitch)); } else if(LastWaterAbsorb > 0.f && !(chanflags&SNDF_NOREVERB)) alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT); else alSourcef(source, AL_PITCH, PITCH(pitch)); if(!reuse_chan) alSourcef(source, AL_SEC_OFFSET, 0.f); else { if((chanflags&SNDF_ABSTIME)) alSourcef(source, AL_SEC_OFFSET, reuse_chan->StartTime.Lo/1000.f); else { // FIXME: set offset based on the current time and the StartTime alSourcef(source, AL_SAMPLE_OFFSET, 0.f); } } if(getALError() != AL_NO_ERROR) return NULL; alSourcei(source, AL_BUFFER, buffer); if((chanflags&SNDF_NOPAUSE) || !SFXPaused) alSourcePlay(source); if(getALError() != AL_NO_ERROR) { alSourcei(source, AL_BUFFER, 0); getALError(); return NULL; } if(!(chanflags&SNDF_NOREVERB)) ReverbSfx.push_back(source); if(!(chanflags&SNDF_NOPAUSE)) PausableSfx.push_back(source); SfxGroup.push_back(source); FreeSfx.pop_back(); FISoundChannel *chan = reuse_chan; if(!chan) chan = S_GetChannel(&source); else chan->SysChannel = &source; chan->Rolloff = *rolloff; chan->DistanceScale = distscale; chan->DistanceSqr = dist_sqr; chan->ManualGain = manualGain; return chan; } void OpenALSoundRenderer::StopChannel(FISoundChannel *chan) { if(chan == NULL || chan->SysChannel == NULL) return; ALuint source = *((ALuint*)chan->SysChannel); // Release first, so it can be properly marked as evicted if it's being // forcefully killed S_ChannelEnded(chan); alSourceRewind(source); alSourcei(source, AL_BUFFER, 0); getALError(); std::vector::iterator i; i = find(PausableSfx.begin(), PausableSfx.end(), source); if(i != PausableSfx.end()) PausableSfx.erase(i); i = find(ReverbSfx.begin(), ReverbSfx.end(), source); if(i != ReverbSfx.end()) ReverbSfx.erase(i); SfxGroup.erase(find(SfxGroup.begin(), SfxGroup.end(), source)); FreeSfx.push_back(source); } unsigned int OpenALSoundRenderer::GetPosition(FISoundChannel *chan) { if(chan == NULL || chan->SysChannel == NULL) return 0; ALint pos; alGetSourcei(*((ALuint*)chan->SysChannel), AL_SAMPLE_OFFSET, &pos); if(getALError() == AL_NO_ERROR) return pos; return 0; } void OpenALSoundRenderer::SetSfxPaused(bool paused, int slot) { int oldslots = SFXPaused; if(paused) { SFXPaused |= 1 << slot; if(oldslots == 0 && PausableSfx.size() > 0) { alSourcePausev(PausableSfx.size(), &PausableSfx[0]); getALError(); PurgeStoppedSources(); } } else { SFXPaused &= ~(1 << slot); if(SFXPaused == 0 && oldslots != 0 && PausableSfx.size() > 0) { alSourcePlayv(PausableSfx.size(), &PausableSfx[0]); getALError(); } } } void OpenALSoundRenderer::SetInactive(bool inactive) { } void OpenALSoundRenderer::Sync(bool sync) { if(sync) { if(SfxGroup.size() > 0) { alSourcePausev(SfxGroup.size(), &SfxGroup[0]); getALError(); PurgeStoppedSources(); } } else { // Might already be something to handle this; basically, get a vector // of all values in SfxGroup that are not also in PausableSfx (when // SFXPaused is non-0). std::vector toplay = SfxGroup; if(SFXPaused) { std::vector::iterator i = toplay.begin(); while(i != toplay.end()) { if(find(PausableSfx.begin(), PausableSfx.end(), *i) != PausableSfx.end()) i = toplay.erase(i); else i++; } } if(toplay.size() > 0) { alSourcePlayv(toplay.size(), &toplay[0]); getALError(); } } } void OpenALSoundRenderer::UpdateSoundParams3D(SoundListener *listener, FISoundChannel *chan, bool areasound, const FVector3 &pos, const FVector3 &vel) { if(chan == NULL || chan->SysChannel == NULL) return; alcSuspendContext(Context); ALuint source = *((ALuint*)chan->SysChannel); alSource3f(source, AL_POSITION, pos[0], pos[1], -pos[2]); alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]); chan->DistanceSqr = (pos - listener->position).LengthSquared() * chan->DistanceScale*chan->DistanceScale; // Not all sources can use the distance models provided by OpenAL. // For the ones that can't, apply the calculated attenuation as the // source gain. Positions still handle the panning, if(chan->ManualGain) { float gain = GetRolloff(&chan->Rolloff, sqrt(chan->DistanceSqr)); alSourcef(source, AL_GAIN, SfxVolume*gain); } getALError(); } void OpenALSoundRenderer::UpdateListener(SoundListener *listener) { if(!listener->valid) return; alcSuspendContext(Context); float angle = listener->angle; ALfloat orient[6]; // forward orient[0] = cos(angle); orient[1] = 0.f; orient[2] = -sin(angle); // up orient[3] = 0.f; orient[4] = 1.f; orient[5] = 0.f; alListenerfv(AL_ORIENTATION, orient); alListener3f(AL_POSITION, listener->position.X, listener->position.Y, -listener->position.Z); alListener3f(AL_VELOCITY, listener->velocity.X, listener->velocity.Y, -listener->velocity.Z); getALError(); const ReverbContainer *env = ForcedEnvironment; if(!env) { env = listener->Environment; if(!env) env = DefaultEnvironments[0]; } if(env != PrevEnvironment || env->Modified) { PrevEnvironment = env; DPrintf("Reverb Environment %s\n", env->Name); if(EnvSlot != 0) LoadReverb(env); const_cast(env)->Modified = false; } // NOTE: Moving into and out of water (and changing water absorption) will // undo pitch variations on sounds if either snd_waterreverb or EFX are // disabled. if(listener->underwater || env->SoftwareWater) { if(LastWaterAbsorb != *snd_waterabsorption) { LastWaterAbsorb = *snd_waterabsorption; if(EnvSlot != 0 && *snd_waterreverb) { // Find the "Underwater" reverb environment env = Environments; while(env && env->ID != 0x1600) env = env->Next; LoadReverb(env ? env : DefaultEnvironments[0]); alFilterf(EnvFilters[0], AL_LOWPASS_GAIN, 0.25f); alFilterf(EnvFilters[0], AL_LOWPASS_GAINHF, 0.75f); alFilterf(EnvFilters[1], AL_LOWPASS_GAIN, 1.f); alFilterf(EnvFilters[1], AL_LOWPASS_GAINHF, 1.f); // Apply the updated filters on the sources foreach(ALuint, i, ReverbSfx) { alSourcef(*i, AL_AIR_ABSORPTION_FACTOR, LastWaterAbsorb); alSourcei(*i, AL_DIRECT_FILTER, EnvFilters[0]); alSource3i(*i, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]); } } else { foreach(ALuint, i, ReverbSfx) alSourcef(*i, AL_PITCH, PITCH_MULT); } getALError(); } } else { if(LastWaterAbsorb > 0.f) { LastWaterAbsorb = 0.f; if(EnvSlot != 0) { LoadReverb(env); alFilterf(EnvFilters[0], AL_LOWPASS_GAIN, 1.f); alFilterf(EnvFilters[0], AL_LOWPASS_GAINHF, 1.f); alFilterf(EnvFilters[1], AL_LOWPASS_GAIN, 1.f); alFilterf(EnvFilters[1], AL_LOWPASS_GAINHF, 1.f); foreach(ALuint, i, ReverbSfx) { alSourcef(*i, AL_AIR_ABSORPTION_FACTOR, 0.f); alSourcei(*i, AL_DIRECT_FILTER, EnvFilters[0]); alSource3i(*i, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]); } } else { foreach(ALuint, i, ReverbSfx) alSourcef(*i, AL_PITCH, 1.f); } getALError(); } } } void OpenALSoundRenderer::UpdateSounds() { alcProcessContext(Context); if(DisconnectNotify) { ALCint connected = ALC_TRUE; alcGetIntegerv(Device, ALC_CONNECTED, 1, &connected); if(connected == ALC_FALSE) { Printf("Sound device disconnected; restarting...\n"); static char snd_reset[] = "snd_reset"; AddCommandString(snd_reset); return; } } PurgeStoppedSources(); } bool OpenALSoundRenderer::IsValid() { return Device != NULL; } void OpenALSoundRenderer::MarkStartTime(FISoundChannel *chan) { // FIXME: Get current time (preferably from the audio clock, but the system // time will have to do) chan->StartTime.AsOne = 0; } float OpenALSoundRenderer::GetAudibility(FISoundChannel *chan) { if(chan == NULL || chan->SysChannel == NULL) return 0.f; ALuint source = *((ALuint*)chan->SysChannel); ALfloat volume = 0.f; if(!chan->ManualGain) volume = SfxVolume * GetRolloff(&chan->Rolloff, sqrt(chan->DistanceSqr)); else { alGetSourcef(source, AL_GAIN, &volume); getALError(); } return volume; } void OpenALSoundRenderer::PrintStatus() { Printf("Output device: "TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_DEVICE_SPECIFIER)); getALCError(Device); ALCint frequency, major, minor, mono, stereo; alcGetIntegerv(Device, ALC_FREQUENCY, 1, &frequency); alcGetIntegerv(Device, ALC_MAJOR_VERSION, 1, &major); alcGetIntegerv(Device, ALC_MINOR_VERSION, 1, &minor); alcGetIntegerv(Device, ALC_MONO_SOURCES, 1, &mono); alcGetIntegerv(Device, ALC_STEREO_SOURCES, 1, &stereo); if(getALCError(Device) == AL_NO_ERROR) { Printf("Device sample rate: "TEXTCOLOR_BLUE"%d"TEXTCOLOR_NORMAL"hz\n", frequency); Printf("ALC Version: "TEXTCOLOR_BLUE"%d.%d\n", major, minor); Printf("ALC Extensions: "TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_EXTENSIONS)); Printf("Available sources: "TEXTCOLOR_BLUE"%d"TEXTCOLOR_NORMAL" ("TEXTCOLOR_BLUE"%d"TEXTCOLOR_NORMAL" mono, "TEXTCOLOR_BLUE"%d"TEXTCOLOR_NORMAL" stereo)\n", mono+stereo, mono, stereo); } if(!alcIsExtensionPresent(Device, "ALC_EXT_EFX")) Printf("EFX not found\n"); else { ALCint sends; alcGetIntegerv(Device, ALC_EFX_MAJOR_VERSION, 1, &major); alcGetIntegerv(Device, ALC_EFX_MINOR_VERSION, 1, &minor); alcGetIntegerv(Device, ALC_MAX_AUXILIARY_SENDS, 1, &sends); if(getALCError(Device) == AL_NO_ERROR) { Printf("EFX Version: "TEXTCOLOR_BLUE"%d.%d\n", major, minor); Printf("Auxiliary sends: "TEXTCOLOR_BLUE"%d\n", sends); } } Printf("Vendor: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VENDOR)); Printf("Renderer: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_RENDERER)); Printf("Version: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VERSION)); Printf("Extensions: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_EXTENSIONS)); getALError(); } FString OpenALSoundRenderer::GatherStats() { ALCint updates = 1; alcGetIntegerv(Device, ALC_REFRESH, 1, &updates); getALCError(Device); ALuint total = Sources.size(); ALuint used = SfxGroup.size()+Streams.size(); ALuint unused = FreeSfx.size(); FString out; out.Format("%u sources ("TEXTCOLOR_YELLOW"%u"TEXTCOLOR_NORMAL" active, "TEXTCOLOR_YELLOW"%u"TEXTCOLOR_NORMAL" free), Update interval: "TEXTCOLOR_YELLOW"%d"TEXTCOLOR_NORMAL"ms", total, used, unused, 1000/updates); return out; } void OpenALSoundRenderer::PrintDriversList() { const ALCchar *drivers = (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") ? alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER) : alcGetString(NULL, ALC_DEVICE_SPECIFIER)); const ALCchar *current = alcGetString(Device, ALC_DEVICE_SPECIFIER); if(drivers == NULL) { Printf(TEXTCOLOR_YELLOW"Failed to retrieve device list: %s\n", alcGetString(NULL, alcGetError(NULL))); return; } Printf("%c%s%2d. %s\n", ' ', ((snd_aldevice=="Default") ? TEXTCOLOR_BOLD : ""), 0, "Default"); for(int i = 1;*drivers;i++) { Printf("%c%s%2d. %s\n", ((strcmp(current, drivers)==0) ? '*' : ' '), ((strcmp(*snd_aldevice, drivers)==0) ? TEXTCOLOR_BOLD : ""), i, drivers); drivers += strlen(drivers)+1; } } void OpenALSoundRenderer::PurgeStoppedSources() { // Release channels that are stopped foreach(ALuint, i, SfxGroup) { ALint state = AL_PLAYING; alGetSourcei(*i, AL_SOURCE_STATE, &state); if(state == AL_PLAYING || state == AL_PAUSED) continue; FSoundChan *schan = Channels; while(schan) { if(schan->SysChannel != NULL && *i == *((ALuint*)schan->SysChannel)) { StopChannel(schan); break; } schan = schan->NextChan; } } getALError(); } void OpenALSoundRenderer::LoadReverb(const ReverbContainer *env) { ALuint &envReverb = EnvEffects[env->ID]; bool doLoad = (env->Modified || !envReverb); if(!envReverb) { bool ok = false; alGenEffects(1, &envReverb); if(getALError() == AL_NO_ERROR) { alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB); ok = (alGetError() == AL_NO_ERROR); if(!ok) { alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB); ok = (alGetError() == AL_NO_ERROR); } if(!ok) { alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_NULL); ok = (alGetError() == AL_NO_ERROR); } if(!ok) { alDeleteEffects(1, &envReverb); getALError(); } } if(!ok) { envReverb = 0; doLoad = false; } } if(doLoad) { const REVERB_PROPERTIES &props = env->Properties; ALint type = AL_EFFECT_NULL; alGetEffecti(envReverb, AL_EFFECT_TYPE, &type); #define mB2Gain(x) ((float)pow(10., (x)/2000.)) if(type == AL_EFFECT_EAXREVERB) { ALfloat reflectpan[3] = { props.ReflectionsPan0, props.ReflectionsPan1, props.ReflectionsPan2 }; ALfloat latepan[3] = { props.ReverbPan0, props.ReverbPan1, props.ReverbPan2 }; #undef SETPARAM #define SETPARAM(e,t,v) alEffectf((e), AL_EAXREVERB_##t, clamp((v), AL_EAXREVERB_MIN_##t, AL_EAXREVERB_MAX_##t)) SETPARAM(envReverb, DENSITY, props.Density/100.f); SETPARAM(envReverb, DIFFUSION, props.Diffusion/100.f); SETPARAM(envReverb, GAIN, mB2Gain(props.Room)); SETPARAM(envReverb, GAINHF, mB2Gain(props.RoomHF)); SETPARAM(envReverb, GAINLF, mB2Gain(props.RoomLF)); SETPARAM(envReverb, DECAY_TIME, props.DecayTime); SETPARAM(envReverb, DECAY_HFRATIO, props.DecayHFRatio); SETPARAM(envReverb, DECAY_LFRATIO, props.DecayLFRatio); SETPARAM(envReverb, REFLECTIONS_GAIN, mB2Gain(props.Reflections)); SETPARAM(envReverb, REFLECTIONS_DELAY, props.ReflectionsDelay); alEffectfv(envReverb, AL_EAXREVERB_REFLECTIONS_PAN, reflectpan); SETPARAM(envReverb, LATE_REVERB_GAIN, mB2Gain(props.Reverb)); SETPARAM(envReverb, LATE_REVERB_DELAY, props.ReverbDelay); alEffectfv(envReverb, AL_EAXREVERB_LATE_REVERB_PAN, latepan); SETPARAM(envReverb, ECHO_TIME, props.EchoTime); SETPARAM(envReverb, ECHO_DEPTH, props.EchoDepth); SETPARAM(envReverb, MODULATION_TIME, props.ModulationTime); SETPARAM(envReverb, MODULATION_DEPTH, props.ModulationDepth); SETPARAM(envReverb, AIR_ABSORPTION_GAINHF, mB2Gain(props.AirAbsorptionHF)); SETPARAM(envReverb, HFREFERENCE, props.HFReference); SETPARAM(envReverb, LFREFERENCE, props.LFReference); SETPARAM(envReverb, ROOM_ROLLOFF_FACTOR, props.RoomRolloffFactor); alEffecti(envReverb, AL_EAXREVERB_DECAY_HFLIMIT, (props.Flags&REVERB_FLAGS_DECAYHFLIMIT)?AL_TRUE:AL_FALSE); #undef SETPARAM } else if(type == AL_EFFECT_REVERB) { #define SETPARAM(e,t,v) alEffectf((e), AL_REVERB_##t, clamp((v), AL_REVERB_MIN_##t, AL_REVERB_MAX_##t)) SETPARAM(envReverb, DENSITY, props.Density/100.f); SETPARAM(envReverb, DIFFUSION, props.Diffusion/100.f); SETPARAM(envReverb, GAIN, mB2Gain(props.Room)); SETPARAM(envReverb, GAINHF, mB2Gain(props.RoomHF)); SETPARAM(envReverb, DECAY_TIME, props.DecayTime); SETPARAM(envReverb, DECAY_HFRATIO, props.DecayHFRatio); SETPARAM(envReverb, REFLECTIONS_GAIN, mB2Gain(props.Reflections)); SETPARAM(envReverb, REFLECTIONS_DELAY, props.ReflectionsDelay); SETPARAM(envReverb, LATE_REVERB_GAIN, mB2Gain(props.Reverb)); SETPARAM(envReverb, LATE_REVERB_DELAY, props.ReverbDelay); SETPARAM(envReverb, AIR_ABSORPTION_GAINHF, mB2Gain(props.AirAbsorptionHF)); SETPARAM(envReverb, ROOM_ROLLOFF_FACTOR, props.RoomRolloffFactor); alEffecti(envReverb, AL_REVERB_DECAY_HFLIMIT, (props.Flags&REVERB_FLAGS_DECAYHFLIMIT)?AL_TRUE:AL_FALSE); #undef SETPARAM } #undef mB2Gain } alAuxiliaryEffectSloti(EnvSlot, AL_EFFECTSLOT_EFFECT, envReverb); getALError(); } FSoundChan *OpenALSoundRenderer::FindLowestChannel() { FSoundChan *schan = Channels; FSoundChan *lowest = NULL; while(schan) { if(schan->SysChannel != NULL) { if(!lowest || schan->Priority < lowest->Priority || (schan->Priority == lowest->Priority && schan->DistanceSqr > lowest->DistanceSqr)) lowest = schan; } schan = schan->NextChan; } return lowest; } #endif // NO_OPENAL