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schoenebeck |
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/*************************************************************************** |
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* * |
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* LinuxSampler - modular, streaming capable sampler * |
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* * |
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* Copyright (C) 2003, 2004 by Benno Senoner and Christian Schoenebeck * |
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* * |
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* This program 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 2 of the License, or * |
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* (at your option) any later version. * |
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* * |
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* This program 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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* * |
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* You should have received a copy of the GNU General Public License * |
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* along with this program; if not, write to the Free Software * |
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, * |
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* MA 02111-1307 USA * |
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***************************************************************************/ |
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#include "AudioOutputDeviceAlsa.h" |
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#include "AudioOutputDeviceFactory.h" |
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namespace LinuxSampler { |
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REGISTER_AUDIO_OUTPUT_DRIVER(AudioOutputDeviceAlsa); |
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/* Common parameters for now they'll have to be registered here. */ |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterActive); |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterSampleRate); |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterChannels); |
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/* Driver specific parameters */ |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterCard); |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragments); |
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REGISTER_AUDIO_OUTPUT_DRIVER_PARAMETER(AudioOutputDeviceAlsa, ParameterFragmentSize); |
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schoenebeck |
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// *************** ParameterCard *************** |
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// * |
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AudioOutputDeviceAlsa::ParameterCard::ParameterCard() : DeviceCreationParameterString() { |
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InitWithDefault(); // use default card |
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} |
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AudioOutputDeviceAlsa::ParameterCard::ParameterCard(String s) throw (LinuxSamplerException) : DeviceCreationParameterString(s) { |
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SetValue(s); // try to use given card |
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} |
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String AudioOutputDeviceAlsa::ParameterCard::Description() { |
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return "Sound card to be used"; |
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} |
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bool AudioOutputDeviceAlsa::ParameterCard::Fix() { |
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return true; |
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} |
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bool AudioOutputDeviceAlsa::ParameterCard::Mandatory() { |
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return false; |
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} |
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std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::ParameterCard::DependsAsParameters() { |
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return std::map<String,DeviceCreationParameter*>(); // no dependencies |
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} |
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optional<String> AudioOutputDeviceAlsa::ParameterCard::DefaultAsString(std::map<String,String> Parameters) { |
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std::vector<String> cards = PossibilitiesAsString(Parameters); |
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if (cards.empty()) throw LinuxSamplerException("AudioOutputDeviceAlsa: Can't find any card"); |
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return cards[0]; // first card by default |
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} |
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std::vector<String> AudioOutputDeviceAlsa::ParameterCard::PossibilitiesAsString(std::map<String,String> Parameters) { |
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int err; |
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std::vector<String> CardNames; |
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// iterate through all cards |
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int card_index = -1; |
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while (snd_card_next(&card_index) >= 0 && card_index >= 0) { |
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String hw_name = "hw:" + ToString(card_index); |
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snd_ctl_t* hCardCtrl; |
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if ((err = snd_ctl_open(&hCardCtrl, hw_name.c_str(), 0)) < 0) { |
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std::cerr << "AudioOutputDeviceAlsa: Cannot open sound control for card " << card_index << " - " << snd_strerror(err) << std::endl; |
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continue; |
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} |
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// iterate through all devices of that card |
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int device_index = -1; |
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while (!snd_ctl_pcm_next_device(hCardCtrl, &device_index) && device_index >= 0) { |
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String name = ToString(card_index) + "," + ToString(device_index); |
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//dmsg(1,("[possibility:%s]", name.c_str())); |
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CardNames.push_back(name); |
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} |
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snd_ctl_close(hCardCtrl); |
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} |
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return CardNames; |
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} |
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void AudioOutputDeviceAlsa::ParameterCard::OnSetValue(String s) throw (LinuxSamplerException) { |
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// not posssible, as parameter is fix |
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} |
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String AudioOutputDeviceAlsa::ParameterCard::Name() { |
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return "card"; |
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} |
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// *************** ParameterFragments *************** |
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// * |
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AudioOutputDeviceAlsa::ParameterFragments::ParameterFragments() : DeviceCreationParameterInt() { |
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InitWithDefault(); |
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} |
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AudioOutputDeviceAlsa::ParameterFragments::ParameterFragments(String s) throw (LinuxSamplerException) : DeviceCreationParameterInt(s) { |
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} |
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String AudioOutputDeviceAlsa::ParameterFragments::Description() { |
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return "Number of buffer fragments"; |
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} |
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bool AudioOutputDeviceAlsa::ParameterFragments::Fix() { |
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return true; |
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} |
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bool AudioOutputDeviceAlsa::ParameterFragments::Mandatory() { |
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return false; |
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} |
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std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::ParameterFragments::DependsAsParameters() { |
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static ParameterCard card; |
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std::map<String,DeviceCreationParameter*> dependencies; |
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dependencies[card.Name()] = &card; |
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return dependencies; |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragments::DefaultAsInt(std::map<String,String> Parameters) { |
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return 2; // until done |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragments::RangeMinAsInt(std::map<String,String> Parameters) { |
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return optional<int>::nothing; |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragments::RangeMaxAsInt(std::map<String,String> Parameters) { |
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return optional<int>::nothing; |
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} |
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std::vector<int> AudioOutputDeviceAlsa::ParameterFragments::PossibilitiesAsInt(std::map<String,String> Parameters) { |
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return std::vector<int>(); |
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} |
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void AudioOutputDeviceAlsa::ParameterFragments::OnSetValue(int i) throw (LinuxSamplerException) { |
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// not posssible, as parameter is fix |
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} |
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String AudioOutputDeviceAlsa::ParameterFragments::Name() { |
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return "fragments"; |
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} |
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// *************** ParameterFragmentSize *************** |
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// * |
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AudioOutputDeviceAlsa::ParameterFragmentSize::ParameterFragmentSize() : DeviceCreationParameterInt() { |
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InitWithDefault(); |
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} |
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AudioOutputDeviceAlsa::ParameterFragmentSize::ParameterFragmentSize(String s) throw (LinuxSamplerException) : DeviceCreationParameterInt(s) { |
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} |
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String AudioOutputDeviceAlsa::ParameterFragmentSize::Description() { |
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return "Size of each buffer fragment"; |
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} |
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bool AudioOutputDeviceAlsa::ParameterFragmentSize::Fix() { |
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return true; |
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} |
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bool AudioOutputDeviceAlsa::ParameterFragmentSize::Mandatory() { |
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return false; |
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} |
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std::map<String,DeviceCreationParameter*> AudioOutputDeviceAlsa::ParameterFragmentSize::DependsAsParameters() { |
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static ParameterCard card; |
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std::map<String,DeviceCreationParameter*> dependencies; |
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dependencies[card.Name()] = &card; |
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return dependencies; |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragmentSize::DefaultAsInt(std::map<String,String> Parameters) { |
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return 128; // until done |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragmentSize::RangeMinAsInt(std::map<String,String> Parameters) { |
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return optional<int>::nothing; |
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} |
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optional<int> AudioOutputDeviceAlsa::ParameterFragmentSize::RangeMaxAsInt(std::map<String,String> Parameters) { |
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return optional<int>::nothing; |
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} |
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std::vector<int> AudioOutputDeviceAlsa::ParameterFragmentSize::PossibilitiesAsInt(std::map<String,String> Parameters) { |
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return std::vector<int>(); |
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} |
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void AudioOutputDeviceAlsa::ParameterFragmentSize::OnSetValue(int i) throw (LinuxSamplerException) { |
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// not posssible, as parameter is fix |
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} |
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String AudioOutputDeviceAlsa::ParameterFragmentSize::Name() { |
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return "fragmentsize"; |
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} |
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// *************** AudioOutputDeviceAlsa *************** |
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// * |
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schoenebeck |
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/** |
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* Create and initialize Alsa audio output device with given parameters. |
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* |
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* @param Parameters - optional parameters |
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* @throws AudioOutputException if output device cannot be opened |
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*/ |
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AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, 1, 0) { |
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pcm_handle = NULL; |
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stream = SND_PCM_STREAM_PLAYBACK; |
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this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["channels"])->ValueAsInt(); |
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this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["samplerate"])->ValueAsInt(); |
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this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["fragmentsize"])->ValueAsInt(); |
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uint Fragments = ((DeviceCreationParameterInt*)Parameters["fragments"])->ValueAsInt(); |
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String Card = Parameters["card"]->Value(); |
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dmsg(1,("Checking if hw parameters supported...\n")); |
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if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) { |
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pcm_name = "hw:" + Card; |
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} |
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else { |
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printf("Warning: your soundcard doesn't support chosen hardware parameters; "); |
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printf("trying to compensate support lack with plughw..."); |
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fflush(stdout); |
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pcm_name = "plughw:" + Card; |
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} |
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dmsg(1,("HW check completed.\n")); |
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int err; |
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snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack. |
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/* Open PCM. The last parameter of this function is the mode. */ |
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/* If this is set to 0, the standard mode is used. Possible */ |
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/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */ |
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/* If SND_PCM_NONBLOCK is used, read / write access to the */ |
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/* PCM device will return immediately. If SND_PCM_ASYNC is */ |
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/* specified, SIGIO will be emitted whenever a period has */ |
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/* been completely processed by the soundcard. */ |
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if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) { |
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throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err)); |
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} |
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if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { |
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throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err)); |
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} |
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/* Set access type. This can be either */ |
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/* SND_PCM_ACCESS_RW_INTERLEAVED or */ |
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/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */ |
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if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { |
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throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err)); |
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} |
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/* Set sample format */ |
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#if WORDS_BIGENDIAN |
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if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0) |
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#else // little endian |
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if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0) |
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#endif |
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{ |
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throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err)); |
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} |
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int dir = 0; |
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/* Set sample rate. If the exact rate is not supported */ |
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/* by the hardware, use nearest possible rate. */ |
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#if ALSA_MAJOR > 0 |
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if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0) |
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#else |
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if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0) |
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#endif |
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{ |
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throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err)); |
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} |
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if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) { |
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throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err)); |
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} |
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/* Set number of periods. Periods used to be called fragments. */ |
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if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) { |
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throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err)); |
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} |
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/* Set buffer size (in frames). The resulting latency is given by */ |
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/* latency = periodsize * periods / (rate * bytes_per_frame) */ |
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if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) { |
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throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err)); |
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} |
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/* Apply HW parameter settings to */ |
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/* PCM device and prepare device */ |
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if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { |
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throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err)); |
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} |
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if (snd_pcm_sw_params_malloc(&swparams) != 0) { |
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throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err)); |
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} |
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if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) { |
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throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err)); |
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} |
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if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) { |
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throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err)); |
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} |
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if (snd_pcm_sw_params(pcm_handle, swparams) != 0) { |
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throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err)); |
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} |
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if ((err = snd_pcm_prepare(pcm_handle)) < 0) { |
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throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err)); |
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} |
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// allocate Alsa output buffer |
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pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize]; |
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// create audio channels for this audio device to which the sampler engines can write to |
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for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(FragmentSize)); |
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if (((DeviceCreationParameterBool*)Parameters["active"])->ValueAsBool()) { |
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Play(); |
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} |
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} |
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AudioOutputDeviceAlsa::~AudioOutputDeviceAlsa() { |
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//dmsg(0,("Stopping Alsa Thread...")); |
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//StopThread(); //FIXME: commented out due to a bug in thread.cpp (StopThread() doesn't return at all) |
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//dmsg(0,("OK\n")); |
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//FIXME: currently commented out due to segfault |
360 |
|
|
//snd_pcm_close(pcm_handle); |
361 |
|
|
|
362 |
|
|
// destroy all audio channels |
363 |
|
|
for (int c = 0; c < Channels.size(); c++) delete Channels[c]; |
364 |
|
|
|
365 |
|
|
if (pAlsaOutputBuffer) { |
366 |
|
|
//FIXME: currently commented out due to segfault |
367 |
|
|
//delete[] pOutputBuffer; |
368 |
|
|
} |
369 |
|
|
} |
370 |
|
|
|
371 |
|
|
/** |
372 |
|
|
* Checks if sound card supports the chosen parameters. |
373 |
|
|
* |
374 |
|
|
* @returns true if hardware supports it |
375 |
|
|
*/ |
376 |
|
|
bool AudioOutputDeviceAlsa::HardwareParametersSupported(String card, uint channels, int samplerate, uint numfragments, uint fragmentsize) { |
377 |
|
|
pcm_name = "hw:" + card; |
378 |
|
|
if (snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0) < 0) return false; |
379 |
|
|
snd_pcm_hw_params_alloca(&hwparams); |
380 |
|
|
if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) { |
381 |
|
|
snd_pcm_close(pcm_handle); |
382 |
|
|
return false; |
383 |
|
|
} |
384 |
|
|
if (snd_pcm_hw_params_test_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { |
385 |
|
|
snd_pcm_close(pcm_handle); |
386 |
|
|
return false; |
387 |
|
|
} |
388 |
|
|
#if WORDS_BIGENDIAN |
389 |
|
|
if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE) < 0) |
390 |
|
|
#else // little endian |
391 |
|
|
if (snd_pcm_hw_params_test_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) |
392 |
|
|
#endif |
393 |
|
|
{ |
394 |
|
|
snd_pcm_close(pcm_handle); |
395 |
|
|
return false; |
396 |
|
|
} |
397 |
|
|
int dir = 0; |
398 |
|
|
if (snd_pcm_hw_params_test_rate(pcm_handle, hwparams, samplerate, dir) < 0) { |
399 |
|
|
snd_pcm_close(pcm_handle); |
400 |
|
|
return false; |
401 |
|
|
} |
402 |
|
|
if (snd_pcm_hw_params_test_channels(pcm_handle, hwparams, channels) < 0) { |
403 |
|
|
snd_pcm_close(pcm_handle); |
404 |
|
|
return false; |
405 |
|
|
} |
406 |
|
|
if (snd_pcm_hw_params_test_periods(pcm_handle, hwparams, numfragments, dir) < 0) { |
407 |
|
|
snd_pcm_close(pcm_handle); |
408 |
|
|
return false; |
409 |
|
|
} |
410 |
|
|
if (snd_pcm_hw_params_test_buffer_size(pcm_handle, hwparams, (fragmentsize * numfragments)) < 0) { |
411 |
|
|
snd_pcm_close(pcm_handle); |
412 |
|
|
return false; |
413 |
|
|
} |
414 |
|
|
|
415 |
|
|
snd_pcm_close(pcm_handle); |
416 |
|
|
return true; |
417 |
|
|
} |
418 |
|
|
|
419 |
|
|
void AudioOutputDeviceAlsa::Play() { |
420 |
|
|
StartThread(); |
421 |
|
|
} |
422 |
|
|
|
423 |
|
|
bool AudioOutputDeviceAlsa::IsPlaying() { |
424 |
|
|
return IsRunning(); // if Thread is running |
425 |
|
|
} |
426 |
|
|
|
427 |
|
|
void AudioOutputDeviceAlsa::Stop() { |
428 |
|
|
StopThread(); |
429 |
|
|
} |
430 |
|
|
|
431 |
|
|
void AudioOutputDeviceAlsa::AcquireChannels(uint Channels) { |
432 |
|
|
if (Channels > uiAlsaChannels) { |
433 |
|
|
// just create mix channel(s) |
434 |
|
|
for (int i = uiAlsaChannels; i < Channels; i++) { |
435 |
|
|
AudioChannel* pNewChannel = new AudioChannel(this->Channels[i % uiAlsaChannels]); |
436 |
|
|
this->Channels.push_back(pNewChannel); |
437 |
|
|
} |
438 |
|
|
} |
439 |
|
|
} |
440 |
|
|
|
441 |
|
|
uint AudioOutputDeviceAlsa::MaxSamplesPerCycle() { |
442 |
|
|
return FragmentSize; |
443 |
|
|
} |
444 |
|
|
|
445 |
|
|
uint AudioOutputDeviceAlsa::SampleRate() { |
446 |
|
|
return uiSamplerate; |
447 |
|
|
} |
448 |
|
|
|
449 |
|
|
String AudioOutputDeviceAlsa::Name() { |
450 |
|
|
return "Alsa"; |
451 |
|
|
} |
452 |
|
|
|
453 |
|
|
String AudioOutputDeviceAlsa::Driver() { |
454 |
|
|
return Name(); |
455 |
|
|
} |
456 |
|
|
|
457 |
|
|
String AudioOutputDeviceAlsa::Description() { |
458 |
|
|
return "Advanced Linux Sound Architecture"; |
459 |
|
|
} |
460 |
|
|
|
461 |
|
|
String AudioOutputDeviceAlsa::Version() { |
462 |
schoenebeck |
212 |
String s = "$Revision: 1.11 $"; |
463 |
schoenebeck |
200 |
return s.substr(11, s.size() - 13); // cut dollar signs, spaces and CVS macro keyword |
464 |
|
|
} |
465 |
|
|
|
466 |
|
|
/** |
467 |
|
|
* Entry point for the thread. |
468 |
|
|
*/ |
469 |
|
|
int AudioOutputDeviceAlsa::Main() { |
470 |
|
|
while (true) { |
471 |
|
|
|
472 |
|
|
// let all connected engines render 'FragmentSize' sample points |
473 |
|
|
RenderAudio(FragmentSize); |
474 |
|
|
|
475 |
|
|
// convert from DSP value range (-1.0..+1.0) to 16 bit integer value |
476 |
|
|
// range (-32768..+32767), check clipping and copy to Alsa output buffer |
477 |
|
|
// (note: we use interleaved output method to Alsa) |
478 |
|
|
for (int c = 0; c < uiAlsaChannels; c++) { |
479 |
|
|
float* in = Channels[c]->Buffer(); |
480 |
|
|
for (int i = 0, o = c; i < FragmentSize; i++ , o += uiAlsaChannels) { |
481 |
|
|
float sample_point = in[i] * 32768.0f; |
482 |
|
|
if (sample_point < -32768.0) sample_point = -32768.0; |
483 |
|
|
if (sample_point > 32767.0) sample_point = 32767.0; |
484 |
|
|
pAlsaOutputBuffer[o] = (int16_t) sample_point; |
485 |
|
|
} |
486 |
|
|
} |
487 |
|
|
|
488 |
|
|
// output sound |
489 |
|
|
int res = Output(); |
490 |
|
|
if (res < 0) { |
491 |
|
|
fprintf(stderr, "Alsa: Audio output error, exiting.\n"); |
492 |
|
|
exit(EXIT_FAILURE); |
493 |
|
|
} |
494 |
|
|
} |
495 |
|
|
} |
496 |
|
|
|
497 |
|
|
/** |
498 |
|
|
* Will be called after every audio fragment cycle, to output the audio data |
499 |
|
|
* of the current fragment to the soundcard. |
500 |
|
|
* |
501 |
|
|
* @returns 0 on success, a value < 0 on error |
502 |
|
|
*/ |
503 |
|
|
int AudioOutputDeviceAlsa::Output() { |
504 |
|
|
int err = snd_pcm_writei(pcm_handle, pAlsaOutputBuffer, FragmentSize); |
505 |
|
|
if (err < 0) { |
506 |
|
|
fprintf(stderr, "Error snd_pcm_writei failed: %s\n", snd_strerror(err)); |
507 |
|
|
return -1; |
508 |
|
|
} |
509 |
|
|
return 0; |
510 |
|
|
} |
511 |
|
|
|
512 |
|
|
} // namespace LinuxSampler |
513 |
|
|
|