diff options
Diffstat (limited to 'PluginRoutingInterfacePulse')
7 files changed, 1960 insertions, 0 deletions
diff --git a/PluginRoutingInterfacePulse/CMakeLists.txt b/PluginRoutingInterfacePulse/CMakeLists.txt new file mode 100644 index 0000000..1a15b40 --- /dev/null +++ b/PluginRoutingInterfacePulse/CMakeLists.txt @@ -0,0 +1,103 @@ +############################################################################ +# SPDX license identifier: MPL-2.0 +# +# Copyright (C) 2012-2014, Wind River Systems +# Copyright (C) 2014, GENIVI Alliance +# +# This file is part of Pulse Audio Interface Routing Plugin. +# +# This Source Code Form is subject to the terms of the Mozilla Public +# License (MPL), v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. +# +# For further information see http://www.genivi.org/. +# +# List of changes: +# +# 21.08.2014, Adrian Scarlat, First version of the code; +# Porting code from AM ver1.x to AM ver3.0; +# Added Copyright and License information; +############################################################################ + +cmake_minimum_required(VERSION 2.6) + +PROJECT(PluginRoutingInterfacePULSE) + +set (CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -rdynamic") +set (CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -g") +FILE(GLOB pulseConf "${CMAKE_CURRENT_SOURCE_DIR}/README*") +FIND_PACKAGE(PkgConfig) + +OPTION( WITH_DOCUMENTATION + "Build together with Doxygen Documentation" OFF ) + +SET(CMAKE_MODULE_PATH ${CMAKE_MODULE_PATH} ${CMAKE_SOURCE_DIR}/cmake) +SET(PLUGINS_OUTPUT_PATH ${CMAKE_SOURCE_DIR}/bin/plugins) +set(LIBRARY_OUTPUT_PATH ${PLUGINS_OUTPUT_PATH}/routing) +set(DOC_OUTPUT_PATH ${CMAKE_SOURCE_DIR}/doc/RoutingPlugin) +set(INCLUDES_FOLDER "include") +SET(LIB_INSTALL_SUFFIX "audioManager") + +FIND_PACKAGE(DBUS REQUIRED) +FIND_PATH(AUDIO_INCLUDE_FOLDER audiomanagertypes.h /usr/include) + +if(DEFINED AUDIO_INCLUDE_FOLDER) + message(STATUS "Found AudioManager include: ${AUDIO_INCLUDE_FOLDER}") +else(DEFINED AUDIO_INCLUDE_FOLDER) + message(STATUS "Did not found AudioManager include!") +endif(DEFINED AUDIO_INCLUDE_FOLDER) + +FILE(READ "${AUDIO_INCLUDE_FOLDER}/routing/IAmRoutingSend.h" VERSION_BUFFER LIMIT 6000) +STRING(REGEX MATCH "RoutingSendVersion*.[^0-9]*[0-9]" LIB_INTERFACE_VERSION_STRING ${VERSION_BUFFER}) +STRING(REGEX REPLACE "[^0-9]" "" LIB_INTERFACE_VERSION ${LIB_INTERFACE_VERSION_STRING}) +MESSAGE(STATUS "Building against routing interface version ${LIB_INTERFACE_VERSION}") + +SET(CMAKE_MODULE_PATH ${CMAKE_MODULE_PATH} ${CMAKE_SOURCE_DIR}/cmake) +FIND_PACKAGE(DBUS REQUIRED) + +INCLUDE_DIRECTORIES( + ${CMAKE_SOURCE_DIR} + ${CMAKE_CURRENT_BINARY_DIR} + ${DBUS_INCLUDE_DIR} + ${DBUS_ARCH_INCLUDE_DIR} + ${AUDIO_INCLUDE_FOLDER} + ${INCLUDES_FOLDER} +) + +# all source files go here +file(GLOB PLUGINDBUS_SRCS_CXX "src/*.cpp") + +add_library(PluginRoutingInterfacePULSE SHARED ${PLUGINDBUS_SRCS_CXX}) + +TARGET_LINK_LIBRARIES(PluginRoutingInterfacePULSE + pulse + ${DLT_LIBRARIES} + ${DBUS_LIBRARY} +) + +IF(WITH_DOCUMENTATION) + file(MAKE_DIRECTORY ${DOC_OUTPUT_PATH}) + configure_file(${DOXY_FILE} ${DOC_OUTPUT_PATH}/Doxyfile @ONLY IMMEDIATE) + add_custom_target (PluginRoutingInterfacePULSEDocs ALL + COMMAND ${DOXYGEN_EXECUTABLE} ${DOC_OUTPUT_PATH}/Doxyfile WORKING_DIRECTORY ${DOC_OUTPUT_PATH} + SOURCES ${PROJECT_BINARY_DIR} ${DOC_OUTPUT_PATH}/Doxyfile + ) +ENDIF(WITH_DOCUMENTATION) + +INSTALL(TARGETS PluginRoutingInterfacePULSE + DESTINATION "lib/${LIB_INSTALL_SUFFIX}/routing" + PERMISSIONS OWNER_EXECUTE OWNER_WRITE OWNER_READ GROUP_EXECUTE GROUP_READ WORLD_EXECUTE WORLD_READ + COMPONENT sampleplugins +) + +INSTALL(FILES data/libPluginRoutingInterfacePULSE.conf + DESTINATION "lib/${LIB_INSTALL_SUFFIX}/routing" + PERMISSIONS OWNER_EXECUTE OWNER_WRITE OWNER_READ GROUP_EXECUTE GROUP_READ WORLD_EXECUTE WORLD_READ + COMPONENT sampleplugins +) + +# Uncomment the following five lines bellow (that start with #CONFIGURE_FILE...) +# to make those files available in build environment; +# For meta-ivi deployment purposes leave them commented. + +#CONFIGURE_FILE( ${CMAKE_SOURCE_DIR}/PluginRoutingInterfacePulse/data/libPluginRoutingInterfacePULSE.conf ${PLUGINS_OUTPUT_PATH}/routing/libPluginRoutingInterfacePULSE.conf ) diff --git a/PluginRoutingInterfacePulse/README b/PluginRoutingInterfacePulse/README new file mode 100644 index 0000000..8238b94 --- /dev/null +++ b/PluginRoutingInterfacePulse/README @@ -0,0 +1,50 @@ +GENIVI_AudioManager_PluginRoutingInterfacePulse +=============================================== +:Author: Adrian Scarlat <adrian.scarlat@windriver.com> +:doctitle: GENIVI_AudioManager_PluginRoutingInterfacePulse + +SPDX license identifier: MPL-2.0 + +Copyright (C) 2011-2014, Wind River Systems +Copyright (C) 2014, GENIVI Alliance + +This file is part of AudioManager Pulse Audio Interface Routing Plugin. + +This Source Code Form is subject to the terms of the Mozilla Public +License (MPL), v. 2.0. If a copy of the MPL was not distributed with this +file, You can obtain one at http://mozilla.org/MPL/2.0/. + +For further information see http://www.genivi.org/. + +== Documentation +Documentation is provided by doxygen. In order to use this, please compile the +AudioManager like this: +cmake -DWITH_DOCUMENTATION=ON +make + +== Description of Pulse Routing Plugin +The PluginRoutingInterfacePulse is used by the AM PoC application to communicate +with PulseAudio present on the system on which the AM PoC application will be +deployed. + +== Build intstructions +Execute the following command from audiomanager/ folder: +mkdir BUILD +cd BUILD +cmake -DWITH_ENABLED_IPC=DBUS -DWITH_PULSE_ROUTING_PLUGIN=ON .. +make [ add to make command "-j 4" if you have a 4 CPU. + +If all goes well a bin/ folder will be made. +Change to it by executing cd ../bin/ + +In order to use the AudioManager with the PluginRoutingInterfacePulse, the +AudioManager must be compiled with PluginRoutingInterfacePulse and with +PluginControlInterfacePulse.For achieving this please consult the README +from the PluginControlInterfacePulse Project also. + +== Available files after building +libPluginRoutingInterfacePULSE.conf -- This is Pulse Routing Interface +configuration files.It is used for configuring Sources and Sinks on the system. +The contend on this file must be in sync with libPluginControlInterface.conf +file from the PluginControlInterfacePulse Projec.Also some PulseAudio knowledge +is advisable. diff --git a/PluginRoutingInterfacePulse/data/libPluginRoutingInterfacePULSE.conf b/PluginRoutingInterfacePulse/data/libPluginRoutingInterfacePULSE.conf new file mode 100644 index 0000000..d8b9868 --- /dev/null +++ b/PluginRoutingInterfacePulse/data/libPluginRoutingInterfacePULSE.conf @@ -0,0 +1,55 @@ +############################################################################ +# SPDX license identifier: MPL-2.0 +# +# Copyright (C) 2012-2014, Wind River Systems +# Copyright (C) 2014, GENIVI Alliance +# +# This file is part of Pulse Audio Interface Routing Plugin. +# +# This Source Code Form is subject to the terms of the Mozilla Public +# License (MPL), v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. +# +# For further information see http://www.genivi.org/. +# +# List of changes: +# +# 21.08.2014, Adrian Scarlat, First version of the code; +# Porting code from AM ver1.x to AM ver3.0; +# Added Copyright and License information; +############################################################################ +# +# config line format: +# +# TYPE|PULSE TYPE|CLASS|NAME|PROPERTY_NAME|PROPERTY_VALUE +# +# TYPE="Source" or "Sink" +# +# PULSE_TYPE="Sink Input" or "Source" for TYPE="Source" +# PULSE_TYPE="Source Output" or "Sink" for TYPE="Sink" +# +# NAME=Any string not containing separator | +# +# CLASS=Any string not containing separator | - name should be consistent with Controller config +# +# PROPERTY_NAME=a Pulse recognized element property, e.g. "application.process.binary" or "device.class" +# for PULSE_TYPE=Source or PULSE_TYPE=Sink, take a look ad device string +# PROPERTY_VALUE=any string corresponding to the property value +# +# TODO: provide support for multiple attributes filtering +# TODO: comment at eof is mandatory! this looks like a bug for the moment +#################### +# Sources +#################### +Source|Sink Input|Entertainment|MediaPlayer|media.role|MEDIA +Source|Sink Input|Navigation|NaviPlayer|media.role|NAVI +Source|Sink Input|TTS|TTSPlayer|media.role|TextToSpeach +Source|Sink Input|Telephony|Skype|media.role|skype +Source|Sink Input|Analogic|ReverseBeep|media.role|reverse +#################### +# Sinks +#################### +Sink|Sink|HifiAudio|AlsaPrimary|na|na +Sink|Sink|HifiAudio|AlsaSecondary|na|na +# !END + diff --git a/PluginRoutingInterfacePulse/include/RoutingSenderMainloopPULSE.h b/PluginRoutingInterfacePulse/include/RoutingSenderMainloopPULSE.h new file mode 100644 index 0000000..ff0c1d8 --- /dev/null +++ b/PluginRoutingInterfacePulse/include/RoutingSenderMainloopPULSE.h @@ -0,0 +1,97 @@ +/** + * SPDX license identifier: MPL-2.0 + * + * Copyright (C) 2011-2014, Wind River Systems + * Copyright (C) 2014, GENIVI Alliance + * + * This file is part of Pulse Audio Interface Routing Plugin. + * + * This Source Code Form is subject to the terms of the Mozilla Public + * License (MPL), v. 2.0. If a copy of the MPL was not distributed with this + * file, You can obtain one at http://mozilla.org/MPL/2.0/. + * + * For further information see http://www.genivi.org/. + * + * List of changes: + * + * 21.08.2014, Adrian Scarlat, First version of the code; + * Porting code from AM ver1.x to AM ver3.0; + * Added Copyright and License information; + */ + +#ifndef ROUTINGSENDERMAINLOPPPULSE_H_ +#define ROUTINGSENDERMAINLOPPPULSE_H_ + +/* Defines */ +/* In PulseAudio the volume value range from 0 (silence) to 0x10000U=65536 ("maximum" sensible volume). */ +#define MAX_PULSE_VOLUME (0x10000U) + +bool routing_sender_create_mainloop(void *thiz); + +void * routing_sender_start_mainloop(void *thiz); + +void routing_sender_get_sink_info_callback( + pa_context *c, + const pa_sink_info *i, + int is_last, void *thiz); + +void routing_sender_context_state_callback(pa_context *c, void *thiz); + +bool routing_sender_get_source_info(pa_context *c, void *thiz); + +void routing_sender_pa_event_callback( + pa_context *c, + pa_subscription_event_type_t t, + uint32_t idx, void *thiz); + +void routing_sender_get_sink_input_info_callback( + pa_context *c, + const pa_sink_input_info *i, + int eol, + void *thiz); + +void routing_sender_get_source_output_info_callback( + pa_context *c, + const pa_source_output_info *i, + int eol, + void *userdata); + +bool routing_sender_move_sink_input( + pa_context *c, + uint32_t sink_input_index, + uint32_t sink_index, + void *thiz); + +bool routing_sender_move_source_output( + pa_context *c, + uint32_t source_output_index, + uint32_t source_index, + void *thiz); + +bool routing_sender_sink_input_volume_ramp( + pa_context *c, + uint32_t sink_input_index, + uint32_t crt_volume, + uint32_t volume, + uint16_t ramp_time, + void *thiz); + +bool routing_sender_sink_input_volume( + pa_context *c, + uint32_t sink_input_index, + uint32_t volume, + void *thiz); + +bool routing_sender_sink_input_mute( + pa_context *c, + uint32_t sink_input_index, + bool mute, + void *thiz); + +bool routing_sender_sink_volume( + pa_context *c, + uint32_t sink_index, + uint32_t volume, + void *thiz); + +#endif diff --git a/PluginRoutingInterfacePulse/include/RoutingSenderPULSE.h b/PluginRoutingInterfacePulse/include/RoutingSenderPULSE.h new file mode 100644 index 0000000..0fc2197 --- /dev/null +++ b/PluginRoutingInterfacePulse/include/RoutingSenderPULSE.h @@ -0,0 +1,131 @@ +/** + * SPDX license identifier: MPL-2.0 + * + * Copyright (C) 2011-2014, Wind River Systems + * Copyright (C) 2014, GENIVI Alliance + * + * This file is part of Pulse Audio Interface Routing Plugin. + * + * This Source Code Form is subject to the terms of the Mozilla Public + * License (MPL), v. 2.0. If a copy of the MPL was not distributed with this + * file, You can obtain one at http://mozilla.org/MPL/2.0/. + * + * For further information see http://www.genivi.org/. + * + * List of changes: + * + * 21.08.2014, Adrian Scarlat, First version of the code; + * Porting code from AM ver1.x to AM ver3.0; + * Added Copyright and License information; + */ + +#ifndef ROUTINGSENDERPULSE_H_ +#define ROUTINGSENDERPULSE_H_ + +/* Includes */ +#include "routing/IAmRoutingSend.h" +#include <signal.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <assert.h> +#include <stdio.h> +#include <stdlib.h> +#include <limits.h> +#include <getopt.h> +#include <locale.h> +#include <map> +#include <pulse/pulseaudio.h> + +using namespace am; + +struct RoutingSenderPULSEConnection +{ + am_connectionID_t connectionID; + am_sourceID_t sourceID; + am_sinkID_t sinkID; + am_Handle_s handle; + bool pending; +}; + + +struct RoutingSenderPULSESourceSinkConfig +{ + am_Source_s source; + am_Sink_s sink; + + std::string name;//e.g. {"gst-launch-0.10", "mono", "aplay"}; + std::string clazz;//e.g. {"Entertainment", "Navigation", "TTS"}; + std::string propertyName;//e.g. {"application.process.binary", "application.process.binary", "application.process.app"}; + std::string propertyValue;// +}; + + +/* Prototypes */ +class RoutingSenderPULSE : public IAmRoutingSend +{ +public: + RoutingSenderPULSE(pa_context *p_paContext); + ~RoutingSenderPULSE(); + + am::am_Error_e startupInterface(am::IAmRoutingReceive* p_routingReceiver); + void setRoutingReady(uint16_t handle); + void setRoutingRundown(uint16_t handle); + am_Error_e asyncAbort(const am_Handle_s handle); + am_Error_e asyncConnect(const am_Handle_s handle, const am_connectionID_t connectionID, const am_sourceID_t sourceID, const am_sinkID_t sinkID, const am_CustomConnectionFormat_t connectionFormat); + am_Error_e asyncDisconnect(const am_Handle_s handle, const am_connectionID_t connectionID); + am_Error_e asyncSetSinkVolume(const am_Handle_s handle, const am_sinkID_t sinkID, const am_volume_t volume, const am_CustomRampType_t ramp, const am_time_t time); + am_Error_e asyncSetSourceVolume(const am_Handle_s handle, const am_sourceID_t sourceID, const am_volume_t volume, const am_CustomRampType_t ramp, const am_time_t time); + am_Error_e asyncSetSourceState(const am_Handle_s handle, const am_sourceID_t sourceID, const am_SourceState_e state); + am_Error_e asyncSetSinkSoundProperties(const am_Handle_s handle, const am_sinkID_t sinkID, const std::vector<am_SoundProperty_s>& listSoundProperties); + am_Error_e asyncSetSinkSoundProperty(const am_Handle_s handle, const am_sinkID_t sinkID, const am_SoundProperty_s& soundProperty); + am_Error_e asyncSetSourceSoundProperties(const am_Handle_s handle, const am_sourceID_t sourceID, const std::vector<am_SoundProperty_s>& listSoundProperties); + am_Error_e asyncSetSourceSoundProperty(const am_Handle_s handle, const am_sourceID_t sourceID, const am_SoundProperty_s& soundProperty); + am_Error_e asyncCrossFade(const am_Handle_s handle, const am_crossfaderID_t crossfaderID, const am_HotSink_e hotSink, const am_CustomRampType_t rampType, const am_time_t time); + am_Error_e setDomainState(const am_domainID_t domainID, const am_DomainState_e domainState); + am_Error_e returnBusName(std::string& BusName) const; + void getInterfaceVersion(std::string& out_ver) const; + + void setPAContext(pa_context *p_paContext) { + this->m_paContext = p_paContext; + } + am_Error_e asyncSetVolumes(const am_Handle_s handle, const std::vector<am_Volumes_s>& listVolumes); + am_Error_e asyncSetSinkNotificationConfiguration(const am_Handle_s handle, const am_sinkID_t sinkID, const am_NotificationConfiguration_s& notificationConfiguration); + am_Error_e asyncSetSourceNotificationConfiguration(const am_Handle_s handle, const am_sourceID_t sourceID, const am_NotificationConfiguration_s& notificationConfiguration); +//Pulse Audio callbacks + void getSinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *userdata); + void getSourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *userdata); + void getSinkInputInfoCallback(pa_context *c, const pa_sink_input_info *i, void *userdata); + void getSourceOutputInfoCallback(pa_context *c, const pa_source_output_info *i, void *userdata); + +private: + void loadConfig(); + + am_Domain_s m_domain; + + std::vector<RoutingSenderPULSESourceSinkConfig> m_sinks; + std::vector<RoutingSenderPULSESourceSinkConfig> m_sources; + + std::map<uint16_t, uint32_t> m_sourceToPASinkInput; + std::map<uint16_t, uint32_t> m_sourceToPASource; + std::map<uint16_t, uint32_t> m_sinkToPASourceOutput; + std::map<uint16_t, uint32_t> m_sinkToPASink; + + uint16_t m_paSinkNullIndex; + uint16_t m_paSourceNullIndex; + + IAmRoutingReceive *m_routingReceiver; + pa_context *m_paContext; + +/** + * Maintain a list of pending actions: there is a high change that the HMI first call connect, + * then the audio client start to play, therefore, sink-input is not yet created by the time "connect" method was called. + * same for volume? not sure - probably the sink input is created when the user change the volume. + * same for disconnect? not sure - probably the sink input was already created by the time the user is calling disconnect + */ + std::vector<RoutingSenderPULSEConnection> m_activeConnections; + std::map<uint16_t, uint16_t> m_sinkToVolume; + std::map<uint16_t, uint16_t> m_sourceToVolume; +}; + +#endif diff --git a/PluginRoutingInterfacePulse/src/RoutingSenderMainloopPULSE.cpp b/PluginRoutingInterfacePulse/src/RoutingSenderMainloopPULSE.cpp new file mode 100644 index 0000000..5738b42 --- /dev/null +++ b/PluginRoutingInterfacePulse/src/RoutingSenderMainloopPULSE.cpp @@ -0,0 +1,610 @@ +/** + * SPDX license identifier: MPL-2.0 + * + * Copyright (C) 2011-2014, Wind River Systems + * Copyright (C) 2014, GENIVI Alliance + * + * This file is part of Pulse Audio Interface Routing Plugin. + * + * This Source Code Form is subject to the terms of the Mozilla Public + * License (MPL), v. 2.0. If a copy of the MPL was not distributed with this + * file, You can obtain one at http://mozilla.org/MPL/2.0/. + * + * For further information see http://www.genivi.org/. + * + * List of changes: + * + * 21.08.2014, Adrian Scarlat, First version of the code; + * Porting code from AM ver1.x to AM ver3.0; + * Added Copyright and License information; + */ + +/* Includes */ + +#include <pthread.h> +#include "shared/CAmDltWrapper.h" +#include "RoutingSenderPULSE.h" +#include "RoutingSenderMainloopPULSE.h" + +static pthread_t *p_thread; +static pa_mainloop *main_loop; + +/* struct used for ramp_volume changing */ +typedef struct ramp_volume +{ + uint32_t sink_input_index; + uint32_t volume_ini; + uint32_t volume_end; + /* max ramp time in ms */ + uint16_t ramp_max_time; + /* current delay between calls in ms */ + uint16_t ramp_crt_elapsed; + /* aux used to calculate delay between calls */ + timespec start_time; +}; +/* map used for storing ramp_volume information for sources */ +std::map<uint32_t, ramp_volume> g_sinkInputId2rampVolume; + +/* Defines */ +#define QUIT_REASON_PA_DOWN 1 +#define QUIT_REASON_AMGR_DOWN 2 + +DLT_IMPORT_CONTEXT(routingPulse) + +bool routing_sender_create_mainloop(void *thiz) +{ + if (!thiz) { + logError("Can not create an working thread for Pulse Audio without a Routing plugin\n"); + return false; + } + p_thread = (pthread_t *) malloc(sizeof(pthread_t)); + pthread_create(p_thread, NULL, routing_sender_start_mainloop, thiz); + return true; +} + + +void* routing_sender_start_mainloop(void *thiz) +{ + pa_mainloop_api *mainloop_api; + pa_proplist *proplist; + pa_context *context; + int ret = 0; + + if (!thiz) { + logError("Can not create an working thread for Pulse Audio without a Routing plugin\n"); + return NULL; + } + + proplist = pa_proplist_new(); + if (pa_proplist_sets(proplist, PA_PROP_APPLICATION_NAME, "RoutingPULSE") < 0) { + logError("Can not prepare Pulse Audio main loop: pa_proplist_sets\n"); + goto end; + } + + if (!(main_loop = pa_mainloop_new())) + { + logError("Can not prepare Pulse Audio main loop: pa_mainloop_new\n"); + goto end; + } + + if (!(mainloop_api = pa_mainloop_get_api(main_loop))) + { + logError("Can not prepare Pulse Audio main loop: pa_mainloop_get_api\n"); + goto end; + } + + if (!(context = pa_context_new_with_proplist(mainloop_api, NULL, proplist))) + { + logError("Can not prepare Pulse Audio main loop: pa_context_new_with_proplist\n"); + goto end; + } + + pa_context_set_state_callback(context, routing_sender_context_state_callback, thiz); + + if (pa_context_connect(context, NULL, pa_context_flags_t(0), NULL) < 0) + { + logError("Can not prepare Pulse Audio main loop: pa_context_new_with_proplist\n"); + goto end; + } + ((RoutingSenderPULSE *) thiz)->setPAContext(context); + + pa_mainloop_run(main_loop, &ret); + +end: + if (ret == QUIT_REASON_PA_DOWN) + { + ret = 0; + //mail loop ended + if (context) + { + pa_context_disconnect(context); + pa_context_unref(context); + context = NULL; + } + + if (proplist) + { + pa_proplist_free(proplist); + proplist = NULL; + } + + if (main_loop) + { + //pa_signal_done(); + pa_signal_done(); + pa_mainloop_free(main_loop); + main_loop = NULL; + } + + //... pulse audio is down ... retry to connect + usleep(100000); + routing_sender_start_mainloop(thiz); + } + //TDOO: else if -check other value for "ret" + else + { + return thiz; + } +} + + +void routing_sender_pa_event_callback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *thiz) +{ + switch(t) + { + case PA_SUBSCRIPTION_EVENT_SINK_INPUT: + { + pa_context_get_sink_input_info( + c, idx, routing_sender_get_sink_input_info_callback, thiz); + break; + } + case PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT: + { + pa_context_get_sink_input_info( + c, idx, routing_sender_get_sink_input_info_callback, thiz); + break; + } + default: + { + logInfo("Pulse Audio event", t, "was ignored"); + } + } +} + + +void routing_sender_get_sink_input_info_callback(pa_context *c, const pa_sink_input_info *i, int eol, void *userdata) +{ + (void) eol; + + RoutingSenderPULSE* thiz = (RoutingSenderPULSE*) userdata; + if (!thiz) + { + logError("pa_context_get_sink_input_info was called with wrong params\n"); + return; + } + + /* init map of ramp_volumes, usefull later */ + if (i != NULL && strcmp(i->name,"null") != 0) + { + ramp_volume source_ramp_volume; + source_ramp_volume.sink_input_index = i->index; + source_ramp_volume.ramp_crt_elapsed = 0; + g_sinkInputId2rampVolume.insert(std::make_pair(i->index, source_ramp_volume)); + } + + thiz->getSinkInputInfoCallback(c, i, userdata); +} + + +void routing_sender_get_source_output_info_callback(pa_context *c, const pa_source_output_info *i, int eol, void *userdata) +{ + (void) eol; + + RoutingSenderPULSE* thiz = (RoutingSenderPULSE*) userdata; + if (!thiz) + { + logError("pa_context_get_surce_output_info was called with wrong params\n"); + return; + } + thiz->getSourceOutputInfoCallback(c, i, userdata); +} + + + +void routing_sender_get_sink_info_callback(pa_context *c, const pa_sink_info *i, int is_last, void *userdata) +{ + RoutingSenderPULSE* thiz = (RoutingSenderPULSE*) userdata; + if (!thiz) + { + logError("pa_context_get_sink_info was called with wrong params\n"); + return; + } + thiz->getSinkInfoCallback(c, i, is_last, userdata); +} + + +void routing_sender_get_source_info_callback(pa_context *c, const pa_source_info *i, int is_last, void *userdata) +{ + RoutingSenderPULSE* thiz = (RoutingSenderPULSE*) userdata; + if (!thiz) + { + logError("pa_context_get_source_info was called with wrong params\n"); + return; + } + thiz->getSourceInfoCallback(c, i, is_last, userdata); +} + + +void routing_sender_subscriber_callback(pa_context *c, int success, void *thiz) { + if (success) + { + pa_operation *o = pa_context_get_sink_info_list(c, routing_sender_get_sink_info_callback, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_get_sink_info_list"); + } + } + else + { + logError("routing_sender_subscriber_callback: success = false"); + } +} + + +void routing_sender_context_state_callback(pa_context *c, void *thiz) + { + + if (pa_context_get_state(c) == PA_CONTEXT_FAILED) + { + pa_mainloop_quit(main_loop, QUIT_REASON_PA_DOWN); + } + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + //when context is ready - subscriber for server events (we are mainly interested in sink inputs & source outputs) + pa_context_set_subscribe_callback(c, routing_sender_pa_event_callback, thiz); + pa_operation *o = pa_context_subscribe(c, PA_SUBSCRIPTION_MASK_ALL, routing_sender_subscriber_callback, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_subscribe"); + } + } + //other states are not relevant +} + + +bool routing_sender_get_source_info(pa_context *c, void *thiz) { + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_operation *o = pa_context_get_source_info_list(c, routing_sender_get_source_info_callback, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_get_sink_info_list"); + return false; + } + } + else + { + logError("Can not get Pulse Audio sources info - context not ready\n"); + return false; + } + + return true; +} + + +bool routing_sender_move_sink_input(pa_context *c, uint32_t sink_input_index, uint32_t sink_index, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_operation *o = pa_context_move_sink_input_by_index(c, sink_input_index, sink_index, NULL, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_move_sink_input_by_index"); + return false; + } + } + else + { + logError("Can not move sink input - context not ready\n"); + return false; + } + return true; +} + + +bool routing_sender_move_source_output(pa_context *c, uint32_t source_output_index, uint32_t source_index, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_operation *o = pa_context_move_source_output_by_index(c, source_output_index, source_index, NULL, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + } + else + { + logError("Can not move source output - context not ready\n"); + return false; + } + return true; +} + +static inline uint16_t timespec2mili(const timespec & time) +{ + return (uint16_t)((time.tv_nsec == -1 && time.tv_sec == -1) ? + -1 : + time.tv_sec * 1000 + time.tv_nsec / 1000000); +} + +/* Considers that time2 > time1 */ +static inline uint16_t timespec2DeltaMili(const timespec & time1, const timespec & time2) +{ + timespec l_deltaTime; + l_deltaTime.tv_sec = time2.tv_sec - time1.tv_sec; + l_deltaTime.tv_nsec = time2.tv_nsec - time1.tv_nsec; + return timespec2mili(l_deltaTime); +} + +static void routing_sender_sink_input_volume_cb(pa_context *c, int success, void *data) +{ + logInfo("routing_sender_sink_input_volume_cb: success=", success, " data=", data); + if (success) + { + ramp_volume * l_ramp_volume = (ramp_volume *)data; + timespec l_endTime; + clock_gettime(0, &l_endTime); + l_ramp_volume->ramp_crt_elapsed = timespec2DeltaMili(l_ramp_volume->start_time, l_endTime); + if (l_ramp_volume->ramp_crt_elapsed >= l_ramp_volume->ramp_max_time) + { + return; + } + + logInfo("routing_sender_sink_input_volume_cb: ms elapsed=", l_ramp_volume->ramp_crt_elapsed, " of ", l_ramp_volume->ramp_max_time); + + /* ######## Calculate new volume with formula: ########## + crt_time x ( vol_end - vol_ini ) + new_vol = vol_ini + ---------------------------------- + max_time + ###################################################### */ + uint32_t new_volume = + l_ramp_volume->volume_ini + + ( ( l_ramp_volume->ramp_crt_elapsed * ( l_ramp_volume->volume_end - l_ramp_volume->volume_ini ) ) / l_ramp_volume->ramp_max_time ); + logInfo("routing_sender_sink_input_volume_cb: vol_ini=",l_ramp_volume->volume_ini,"vol_crt=",new_volume,"vol_end=",l_ramp_volume->volume_end); + + + /* ***** Set volume again ***** */ + pa_cvolume *volumeCh = (pa_cvolume *) malloc(sizeof(pa_cvolume)); + volumeCh->channels = 1;//TODO: check is stream is mono / stereo + volumeCh->values[0] = new_volume; + + pa_operation *o = pa_context_set_sink_input_volume(c, l_ramp_volume->sink_input_index, volumeCh, NULL, NULL); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return ; + } + usleep(10000); + + volumeCh->channels = 2;//TODO: check is stream is mono / stereo + volumeCh->values[0] = new_volume; + volumeCh->values[1] = new_volume; + logInfo("routing_sender_sink_input_volume_cb: will set vol=", new_volume); + o = pa_context_set_sink_input_volume(c, l_ramp_volume->sink_input_index, volumeCh, routing_sender_sink_input_volume_cb, l_ramp_volume); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return ; + } + } + +} + +bool routing_sender_sink_input_volume_ramp(pa_context *c, uint32_t sink_input_index, uint32_t crt_volume, uint32_t volume, uint16_t ramp_time, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + /* before everything, check to see if ramp_volume struct exists */ + std::map<uint32_t, ramp_volume>::iterator iter = g_sinkInputId2rampVolume.find(sink_input_index); + std::map<uint32_t, ramp_volume>::iterator iterEnd = g_sinkInputId2rampVolume.end(); + if (iter != iterEnd) + { + /* set test volume with only 1 unit more or less, just to see how callback responds */ + pa_volume_t test_volume = ((crt_volume * MAX_PULSE_VOLUME) / 100); + test_volume += volume > crt_volume ? 1 : -1; + + pa_cvolume *volumeCh = (pa_cvolume *) malloc(sizeof(pa_cvolume)); + volumeCh->channels = 1;//TODO: check is stream is mono / stereo + volumeCh->values[0] = test_volume; + + pa_operation *o = pa_context_set_sink_input_volume(c, sink_input_index, volumeCh, NULL, NULL); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + + volumeCh->channels = 2;//TODO: check is stream is mono / stereo + volumeCh->values[0] = test_volume; + volumeCh->values[1] = test_volume; + + ramp_volume * l_ramp_volume = (ramp_volume *) &iter->second; + logInfo("routing_sender_sink_input_volume_ramp: searching ",sink_input_index,"found ramp_vlume struct with sinkInputId ",l_ramp_volume->sink_input_index," (should be equal) "); + l_ramp_volume->volume_ini = (crt_volume * MAX_PULSE_VOLUME) / 100; + l_ramp_volume->volume_end = (volume * MAX_PULSE_VOLUME) / 100; + l_ramp_volume->ramp_max_time = ramp_time; + clock_gettime(0, &l_ramp_volume->start_time); + l_ramp_volume->ramp_crt_elapsed = 0; + logInfo("routing_sender_sink_input_volume_ramp: will set vol=", test_volume); + o = pa_context_set_sink_input_volume(c, sink_input_index, volumeCh, routing_sender_sink_input_volume_cb, l_ramp_volume); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + } + else + { + logInfo("routing_sender_sink_input_volume_ramp: didn't find struct with sinkInputId ", sink_input_index); + /* make-it the old traditional way */ + return routing_sender_sink_input_volume(c, sink_input_index, volume, thiz); + } + + + } + else + { + logError("Can not set sink input volume - context not ready\n"); + return false; + } + return true; +} + +bool routing_sender_sink_input_volume(pa_context *c, uint32_t sink_input_index, uint32_t volume, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_cvolume *volumeCh = (pa_cvolume *) malloc(sizeof(pa_cvolume)); + volumeCh->channels = 1;//TODO: check is stream is mono / stereo + volumeCh->values[0] = (volume * MAX_PULSE_VOLUME) / 100; + + pa_operation *o = pa_context_set_sink_input_volume(c, sink_input_index, volumeCh, NULL, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + + volumeCh->channels = 2;//TODO: check is stream is mono / stereo + volumeCh->values[0] = (volume * MAX_PULSE_VOLUME) / 100; + volumeCh->values[1] = (volume * MAX_PULSE_VOLUME) / 100; + + o = pa_context_set_sink_input_volume(c, sink_input_index, volumeCh, NULL, NULL); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + } + else + { + logError("Can not set sink input volume - context not ready\n"); + return false; + } + return true; +} + + +bool routing_sender_sink_input_mute(pa_context *c, uint32_t sink_input_index, bool mute, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_operation *o = pa_context_set_sink_input_mute( + c, sink_input_index, mute ? 1 : 0, NULL, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_mute"); + return false; + } + } + else + { + logError("Can not set sink input volume - context not ready\n"); + return false; + } + return true; +} + + +bool routing_sender_sink_volume(pa_context *c, uint32_t sink_index, uint32_t volume, void *thiz) +{ + if (pa_context_get_state(c) == PA_CONTEXT_READY) + { + pa_cvolume *volumeCh = (pa_cvolume *) malloc(sizeof(pa_cvolume)); + volumeCh->channels = 2;//TODO: check is stream is mono / stereo + volumeCh->values[0] = (volume * MAX_PULSE_VOLUME) / 100; + volumeCh->values[1] = (volume * MAX_PULSE_VOLUME) / 100; + + pa_operation *o = pa_context_set_sink_volume_by_index(c, sink_index, volumeCh, NULL, thiz); + if (o) + { + pa_operation_unref(o); + } + else + { + logError("Unable to create Pulse Audio operation:", + "pa_context_set_sink_input_volume"); + return false; + } + } + else + { + logError("Can not set sink input volume - context not ready\n"); + return false; + } + return true; +} + +//TODO - implements mute/un-mute and sink suspend(pause) +//TODO - IMPORTANT !! implement volume change for sink input even multiple sink inputs are created during one connection diff --git a/PluginRoutingInterfacePulse/src/RoutingSenderPULSE.cpp b/PluginRoutingInterfacePulse/src/RoutingSenderPULSE.cpp new file mode 100644 index 0000000..17422e9 --- /dev/null +++ b/PluginRoutingInterfacePulse/src/RoutingSenderPULSE.cpp @@ -0,0 +1,914 @@ +/** + * SPDX license identifier: MPL-2.0 + * + * Copyright (C) 2011-2014, Wind River Systems + * Copyright (C) 2014, GENIVI Alliance + * + * This file is part of Pulse Audio Interface Routing Plugin. + * + * This Source Code Form is subject to the terms of the Mozilla Public + * License (MPL), v. 2.0. If a copy of the MPL was not distributed with this + * file, You can obtain one at http://mozilla.org/MPL/2.0/. + * + * For further information see http://www.genivi.org/. + * + * List of changes: + * + * 21.08.2014, Adrian Scarlat, First version of the code; + * Porting code from AM ver1.x to AM ver3.0; + * Added Copyright and License information; + * + * + * DESCRIPTION + * + * This module is handling requests form AudioManager daemon and redirect them to Pulse Audio server. + * It keeps track of existing audio sink, sources, sink-input, sources-input and performs connection/disconnection + * and other operations upon AudioManager daemon request. + * + * The modul is configured with a static list of sink and sources. + * Sinks are: audio output(e.g. speakers) or recording applications. + * Sources are: playback applications or audio input(e.g. microphone) + * + * In PulseAudio server those are classified as follows: + * - audio output - sinks: identified by name or properties: device.api (e.g = "alsa") & device.class(e.g. = "sound") + * - audio input - sources: identified by name or properties: device.api (e.g = "alsa") & device.class(e.g. = "sound") + * - playback applications - sink inputs: identified by application.name (e.g. "ALSA plug-in [chromium-browser]", + * - recording applications - source outputs: identified by application.name (e.g. "ALSA plug-in [chromium-browser]", + * application.process.user = "popai", application.process.binary = "chromium-browser") + * + */ + +#include <stdio.h> +#include <string.h> +#include <iostream> + +#include "shared/CAmDltWrapper.h" + +#include "RoutingSenderPULSE.h" +#include "RoutingSenderMainloopPULSE.h" + + + +#define LIBNAME "libPluginRoutingInterfacePULSE.so" +#define CFGNAME "libPluginRoutingInterfacePULSE.conf" + +/* Globals */ + + +/* Defines */ +DLT_DECLARE_CONTEXT(routingPulse) +/* Maximum source volume measured in percentage. Minimum value is 0% */ +#define MAX_SOURCE_VOLUME (100) + + +/** + * Factory function for the plug-in to be used by audio manager daemon with dlopen & dlsym functions. + * Pattern "libraryName"FACTORY + * + * @author Ionut Popa (ionut.popa@windriver.com) + * @return an instance of RoutingSendInterface or type RoutingSenderPULSE. + */ +extern "C" IAmRoutingSend* PluginRoutingInterfacePULSEFactory() +{ + + return (new RoutingSenderPULSE(NULL)); +} + +/** + * Destructor function for the plug-in to be used by audio manager daemon with dl_open & dl_sym functions. + * + * @param routingSendInterface - the instance created by PluginRoutingInterfaceDbusFactory + * @author Ionut Popa (ionut.popa@windriver.com) + */ +extern "C" void destroyPluginRoutingInterfacePULSE(IAmRoutingSend* routingSendInterface) +{ + delete routingSendInterface;//virtual destructor -> our constructor will be called too +} + + +/** + * Constructor. + * @param p_paContext - reference to PulseAudio context + */ +RoutingSenderPULSE::RoutingSenderPULSE(pa_context *p_paContext) +{ + this->m_paSinkNullIndex = -1; + this->m_paSourceNullIndex = -1; + this->m_paContext = p_paContext; +} + + +void RoutingSenderPULSE::loadConfig() +{ + //get current library path - search: /proc/< getpid() >/maps + char proc_maps_file_name[256]; + char line[256]; + char lib_name[256]; + char *tmp; + pid_t pid = getpid(); + snprintf(proc_maps_file_name, 256, "/proc/%d/maps", pid); + FILE *proc_maps = fopen(proc_maps_file_name, "r"); + + while (!feof(proc_maps)) + { + char *cnt = fgets(line, 256, proc_maps); + if (strlen(line) == 0 || line[0] == '#') + { + continue; + } + if (cnt == NULL) continue; + //tmp0 tmp1 tmp2 tmp3 tmp4 lib_name); + tmp = strtok(line, " ");//address-interval + if(tmp == NULL) continue; + + tmp = strtok(NULL, " ");//rights + if(tmp == NULL) continue; + + tmp = strtok(NULL, " ");//offset + if(tmp == NULL) continue; + + strtok(NULL, " ");//dev + if(tmp == NULL) continue; + + tmp = strtok(NULL, " \n");//inode + if(tmp == NULL) continue; + + tmp = strtok(NULL, " \n"); + if(tmp == NULL) continue; + + strcpy(lib_name, tmp); + if ((lib_name != NULL) && (strstr(lib_name, LIBNAME) >= lib_name)) + { + strcpy(strrchr(lib_name, '/') + 1, CFGNAME); + logInfo("PULSE - config file name: %s\n", lib_name); + + FILE *config = fopen(lib_name, "r"); + + while (config && !feof(config)) + { + char *cnt = fgets(line, 256, config); + if (!line || strlen(line) == 0) continue; + //config format line: TYPE|PULSE TYPE|NAME|CLASS|PROPERTY_NAME|PROPERTY_VALUE + //TYPE="Source" or "Sink" + + char *tmp = strtok(line, "|");//type + if (strcmp("Sink", tmp) == 0) + { + //add sink config + RoutingSenderPULSESourceSinkConfig sinkConfig; + + tmp = strtok(NULL, "|");//pulse type - not used for the moment + + tmp = strtok(NULL, "|");//class + sinkConfig.clazz = std::string(tmp); + + tmp = strtok(NULL, "|");//name + sinkConfig.name = std::string(tmp); + + tmp = strtok(NULL, "|");//property name + sinkConfig.propertyName = std::string(tmp); + + tmp = strtok(NULL, "|\n");//property value + sinkConfig.propertyValue = std::string(tmp); + + m_sinks.push_back(sinkConfig); + logInfo("sinkConfig: sinkConfig.clazz=", sinkConfig.clazz, " sinkConfig.name=", sinkConfig.name, " sinkConfig.propertyName=", sinkConfig.propertyName, " sinkConfig.propertyValue=", sinkConfig.propertyValue); + } + if (strcmp("Source", tmp) == 0) + { + //add source config + RoutingSenderPULSESourceSinkConfig sourceConfig; + + tmp = strtok(NULL, "|");//pulse type - not used for the moment + + tmp = strtok(NULL, "|");//class + sourceConfig.clazz = std::string(tmp); + + tmp = strtok(NULL, "|");//name + sourceConfig.name = std::string(tmp); + + tmp = strtok(NULL, "|");//property name + sourceConfig.propertyName = std::string(tmp); + + tmp = strtok(NULL, "|\n");//property value + sourceConfig.propertyValue = std::string(tmp); + + m_sources.push_back(sourceConfig); + logInfo("sourceConfig: sourceConfig.clazz=", sourceConfig.clazz, " sourceConfig.name=", sourceConfig.name, " sourceConfig.propertyName=", sourceConfig.propertyName, " sourceConfig.propertyValue=", sourceConfig.propertyValue); + } + } + + if (config) + fclose(config); + break; + } + } + + fclose(proc_maps); +} + + +/** + * Destructor. + */ +RoutingSenderPULSE::~RoutingSenderPULSE() +{ + //TODO: Disconnect from pulse: quit main loop and free the context and stuff +} + +/** + * Connecting sender & receiver + * @author Ionut Popa (ionut.popa@windriver.com) + */ +am_Error_e RoutingSenderPULSE::startupInterface(am::IAmRoutingReceive *p_routingReceiver) +{ + this->m_routingReceiver = p_routingReceiver; + return am::E_OK; +} + +void RoutingSenderPULSE::setRoutingReady(uint16_t handle) +{ + //TODO: do not register sinks with the same name + + int i; + this->loadConfig(); + //first register Domain = PulseAudio + this->m_domain.name = "PulseAudio"; + this->returnBusName(this->m_domain.busname);//set domain bus name = current interface bus name + this->m_domain.nodename = "PulseAudio"; + this->m_domain.early = false; + this->m_domain.complete = true; + this->m_domain.state = am::DS_CONTROLLED; + + this->m_domain.domainID = 0; + this->m_routingReceiver->registerDomain(this->m_domain, this->m_domain.domainID); + + am_SoundProperty_s l_spTreble; + l_spTreble.type = SP_GENIVI_BASS; + l_spTreble.value = 0; + + am_SoundProperty_s l_spMid; + l_spMid.type = SP_GENIVI_MID; + l_spMid.value = 0; + + am_SoundProperty_s l_spBass; + l_spBass.type = SP_GENIVI_BASS; + l_spBass.value = 0; + + //register sources (sink inputs & sinks) + for (i = 0; i < m_sources.size(); i++) + { + am_sourceID_t l_newSourceID = 0; + this->m_sources[i].source.sourceID = l_newSourceID; + this->m_sources[i].source.name = m_sources[i].name; + this->m_sources[i].source.sourceState = am::SS_ON; + this->m_sources[i].source.domainID = this->m_domain.domainID; + this->m_sources[i].source.visible = true; + this->m_sources[i].source.volume = MAX_SOURCE_VOLUME; /* initialize source volume to 100% */ + + this->m_sources[i].source.listConnectionFormats.push_back(am::CF_GENIVI_STEREO); + this->m_routingReceiver->peekSourceClassID( + this->m_sources[i].clazz, + this->m_sources[i].source.sourceClassID); + + this->m_routingReceiver->registerSource(this->m_sources[i].source, l_newSourceID); + + this->m_sources[i].source.sourceID = l_newSourceID; + m_sourceToPASinkInput[l_newSourceID] = -1; + m_sourceToPASource[l_newSourceID] = -1; + + logInfo("PULSE - register source:" + ,m_sources[i].name + , "(", m_sources[i].propertyName , ", ", m_sources[i].propertyValue, ")"); + m_sourceToVolume[l_newSourceID] = MAX_SOURCE_VOLUME;//initially all the sources are at 100% + } + + //register sinks (source outputs & sources) + for (i = 0; i < m_sinks.size(); i++) + { + am_sinkID_t l_newsinkID = 0; + this->m_sinks[i].sink.sinkID = l_newsinkID; + this->m_sinks[i].sink.name = this->m_sinks[i].name; + this->m_sinks[i].sink.muteState = am::MS_MUTED; + this->m_sinks[i].sink.domainID = this->m_domain.domainID; + this->m_sinks[i].sink.visible = true; + + this->m_sinks[i].sink.listSoundProperties.push_back(l_spTreble); + this->m_sinks[i].sink.listSoundProperties.push_back(l_spMid); + this->m_sinks[i].sink.listSoundProperties.push_back(l_spBass); + this->m_sinks[i].sink.listConnectionFormats.push_back(am::CF_GENIVI_STEREO); + + this->m_routingReceiver->peekSinkClassID( + this->m_sinks[i].clazz, + this->m_sinks[i].sink.sinkClassID); + this->m_routingReceiver->registerSink(this->m_sinks[i].sink, l_newsinkID); + this->m_sinks[i].sink.sinkID = l_newsinkID; + m_sinkToPASourceOutput[l_newsinkID] = -1; + m_sinkToPASink[l_newsinkID] = -1; + + logInfo("PULSE - register sink:" + ,m_sinks[i].name + , "(", m_sinks[i].propertyName , ", ", m_sinks[i].propertyValue, ")"); + } + + logInfo("PULSE - routingInterfacesReady"); + this->m_routingReceiver->confirmRoutingReady(handle, am::E_OK); + + //register pulse sink & sources, sink inputs & source outputs - > start the main PA loop + routing_sender_create_mainloop((void *) this); +} + +void RoutingSenderPULSE::setRoutingRundown(uint16_t handle) +{ + this->m_routingReceiver->confirmRoutingRundown(handle, am::E_OK); + //TODO: implement this +} + +am_Error_e RoutingSenderPULSE::asyncAbort(const am_Handle_s handle) +{ + (void) handle; + return E_NOT_USED; +} + + +am_Error_e RoutingSenderPULSE::asyncConnect( + const am_Handle_s handle, + const am_connectionID_t connectionID, + const am_sourceID_t sourceID, + const am_sinkID_t sinkID, + const am_CustomConnectionFormat_t connectionFormat) +{ + //TODO: check stuff like connectionFormat + logInfo("PULSE - asyncConnect() - start"); + //add source,sink & connectionID to a list of connections maintained by Routing Pulse Engine + RoutingSenderPULSEConnection l_newConnection; + l_newConnection.sinkID = sinkID; + l_newConnection.sourceID = sourceID; + l_newConnection.connectionID = connectionID; + l_newConnection.handle = handle; + + //by default - sources ar connected at 100% -> controller is responsible to setSourcevolume if needed + + m_sourceToVolume[sourceID] = MAX_SOURCE_VOLUME; + + + if (m_sinkToPASink[sinkID] != -1) + { + if (m_sourceToPASinkInput[sourceID] != -1) + { + if (routing_sender_move_sink_input( + this->m_paContext, + m_sourceToPASinkInput[sourceID], + m_sinkToPASink[sinkID], + this)) + { + //TODO: add callback for pulse move sink input -> to send confirmation; for the moment directly send confirmation + + logInfo("PULSE - asyncConnect() - connectionID:", connectionID, + "move sinkInputIndex:", m_sourceToPASinkInput[sourceID], "to sinkIndex:", m_sinkToPASink[sinkID]); + } + else + { + this->m_routingReceiver->ackConnect(handle, connectionID, am::E_NOT_POSSIBLE); + return am::E_NOT_POSSIBLE; + } + }//else move_sink_input will be called later + } + else if (m_sourceToPASource[sourceID] != -1) + { + if (m_sinkToPASourceOutput[sinkID] != -1) + { + if (routing_sender_move_source_output( + this->m_paContext, + m_sinkToPASourceOutput[sinkID], + m_sourceToPASource[sourceID], + this)) + { + //TODO: add callback for pulse move sink input -> to send confirmation; for the moment directly send confirmation + + logInfo("PULSE - asyncConnect() - connectionID:", connectionID, + "move sourceOutputIndex:", m_sinkToPASourceOutput[sinkID], "to sourceIndex:", m_sourceToPASource[sourceID]); + + } + else + { + this->m_routingReceiver->ackConnect(handle, connectionID, am::E_NOT_POSSIBLE); + return am::E_NOT_POSSIBLE; + } + }//else move_sink_input will be called later + } + else + { + logError("Sink and source for connection not identified:", + sinkID, sourceID, connectionID); + return am::E_NOT_POSSIBLE; + } + + m_activeConnections.push_back(l_newConnection); + + this->m_routingReceiver->ackConnect(handle, connectionID, am::E_OK); + +/** + * TODO: connection is always possible ? check that +*/ + + return am::E_OK; +} + + +am_Error_e RoutingSenderPULSE::asyncDisconnect(const am_Handle_s handle, const am_connectionID_t connectionID) +{ + //get connection by ID ... not to many connections, therefore linear search is fast enough + std::vector<RoutingSenderPULSEConnection>::iterator iter = m_activeConnections.begin(); + std::vector<RoutingSenderPULSEConnection>::iterator iterEnd = m_activeConnections.end(); + for (; iter < iterEnd; ++iter) + { + if (iter->connectionID == connectionID) + { + if (m_sourceToPASinkInput[iter->sourceID] != -1) + { + if (this->m_paSinkNullIndex >= 0) + { + //if null sink is defined - disconnect = move sink input to null + logInfo("PULSE - asyncDisconnect() - connection found - move sinkInputIndex:", + m_sourceToPASinkInput[iter->sourceID], "to NULL sinkIndex:", this->m_paSinkNullIndex); + + routing_sender_move_sink_input( + this->m_paContext, + this->m_sourceToPASinkInput[iter->sourceID], + this->m_paSinkNullIndex, + this); + //TODO: add callback for pulse move sink input -> to send confirmation; for the moment directly send confirmation + this->m_routingReceiver->ackDisconnect(handle, connectionID, am::E_OK); + + } + } + else if (m_sinkToPASourceOutput[iter->sinkID] != -1) + { + if (this->m_paSourceNullIndex >= 0) + { + //if null source is defined - disconnect = move source output to null + logInfo("PULSE - asyncDisconnect() - connection found - move sourceOutputIndex:", + m_sinkToPASourceOutput[iter->sinkID], "to NULL sourceIndex:", this->m_paSourceNullIndex); + + routing_sender_move_source_output( + this->m_paContext, + m_sourceToPASinkInput[iter->sourceID], + this->m_paSinkNullIndex, + this); + //TODO: add callback for pulse move sink input -> to send confirmation; for the moment directly send confirmation + this->m_routingReceiver->ackDisconnect(handle, connectionID, am::E_OK); + + //remove connection from the list of active connections + iter = m_activeConnections.erase(iter); + + break; + } + } + else + { + logInfo("PULSE - asyncDisconnect() - connection found - but no sink input or source"); + this->m_routingReceiver->ackDisconnect(handle, connectionID, am::E_OK); + } + //remove connection from the list of active connections + iter = m_activeConnections.erase(iter); + + break; + } + } + return am::E_OK; +} + +am_Error_e RoutingSenderPULSE::asyncSetSinkVolume( + const am_Handle_s handle, + const am_sinkID_t sinkID, + const am_volume_t volume, + const am_CustomRampType_t ramp, + const am_time_t time) +{ + (void) ramp; + (void) time; + + logInfo("PULSE - asyncSetSinkVolume() - volume:", volume, "sink index:", this->m_sinkToPASink[sinkID]); + + routing_sender_sink_volume( + this->m_paContext, + this->m_sinkToPASink[sinkID], + volume, + this); + this->m_routingReceiver->ackSetSinkVolumeChange(handle, volume, E_OK); + return E_OK; +} + +am_Error_e RoutingSenderPULSE::asyncSetSourceVolume( + const am_Handle_s handle, + const am_sourceID_t sourceID, + const am_volume_t volume, + const am_CustomRampType_t ramp, + const am_time_t time) +{ + (void) ramp; + (void) time; + + am_volume_t crt_volume = this->m_sourceToVolume[sourceID]; + this->m_sourceToVolume[sourceID] = volume; + + logInfo("PULSE - asyncSetSourceVolume() - volume:", volume, "sink input index:", this->m_sourceToPASinkInput[sourceID]); + if (m_sourceToPASinkInput[sourceID] != -1) + { + if (time == 0) + {/* without ramp time */ + routing_sender_sink_input_volume( + this->m_paContext, + this->m_sourceToPASinkInput[sourceID], + volume, + this); + } + else + {/* with ramp time */ + routing_sender_sink_input_volume_ramp( + this->m_paContext, + this->m_sourceToPASinkInput[sourceID], + crt_volume, + volume, + (uint16_t)time, + this); + } + } + else + { + logInfo("PULSE - sink input not registered yet - should wait for registration before update the volume"); + } + this->m_routingReceiver->ackSetSourceVolumeChange(handle, volume, E_OK); + +} + +am_Error_e RoutingSenderPULSE::asyncSetSourceState( + const am_Handle_s handle, + const am_sourceID_t sourceID, + const am_SourceState_e state) +{ + logInfo("PULSE - asyncSetSourceState", state); + switch (state) + { + case SS_ON: + { + routing_sender_sink_input_mute( + this->m_paContext, + this->m_sourceToPASinkInput[sourceID], + false, + this + ); + break; + } + case SS_OFF: + case SS_PAUSED: + { + //TODO: mute source in case of PAUSE or OFF - is there a better way to pause ? maybe suspending the associated sink? + routing_sender_sink_input_mute( + this->m_paContext, + this->m_sourceToPASinkInput[sourceID], + true, + this + ); + break; + } + default: + { + logError("RoutingSenderPULSE::asyncSetSourceState - wrong source state\n"); + this->m_routingReceiver->ackSetSourceState(handle, E_NOT_POSSIBLE); + return E_NOT_POSSIBLE; + } + } + this->m_routingReceiver->ackSetSourceState(handle, E_OK); + return E_OK; +} + +am_Error_e RoutingSenderPULSE::asyncSetSinkSoundProperties( + const am_Handle_s handle, + const am_sinkID_t sinkID, + const std::vector<am_SoundProperty_s>& listSoundProperties) +{ + (void) handle; + (void) sinkID; + (void) listSoundProperties; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::asyncSetSinkSoundProperty( + const am_Handle_s handle, + const am_sinkID_t sinkID, + const am_SoundProperty_s& soundProperty) +{ + (void) handle; + (void) sinkID; + (void) soundProperty; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::asyncSetSourceSoundProperties( + const am_Handle_s handle, + const am_sourceID_t sourceID, + const std::vector<am_SoundProperty_s>& listSoundProperties) +{ + (void) handle; + (void) sourceID; + (void) listSoundProperties; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::asyncSetSourceSoundProperty( + const am_Handle_s handle, + const am_sourceID_t sourceID, + const am_SoundProperty_s& soundProperty) +{ + (void) handle; + (void) sourceID; + (void) soundProperty; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::asyncCrossFade( + const am_Handle_s handle, + const am_crossfaderID_t crossfaderID, + const am_HotSink_e hotSink, + const am_CustomRampType_t rampType, + const am_time_t time) +{ + (void) handle; + (void) crossfaderID; + (void) hotSink; + (void) rampType; + (void) time; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::setDomainState( + const am_domainID_t domainID, + const am_DomainState_e domainState) +{ + (void) domainID; + (void) domainState; + return E_NOT_USED; +} + +am_Error_e RoutingSenderPULSE::returnBusName(std::string& BusName) const { + BusName = "RoutingPULSE"; + return E_OK; +} + +void RoutingSenderPULSE::getInterfaceVersion(std::string& out_ver) const +{ + out_ver = "3.0"; +} + +am_Error_e RoutingSenderPULSE::asyncSetVolumes(const am_Handle_s handle, const std::vector<am_Volumes_s>& listVolumes) +{ + (void) handle; + (void) listVolumes; + //todo: implement asyncSetVolumes; + return (E_NOT_USED); +} + +am_Error_e RoutingSenderPULSE::asyncSetSinkNotificationConfiguration(const am_Handle_s handle, const am_sinkID_t sinkID, const am_NotificationConfiguration_s& notificationConfiguration) +{ + (void) handle; + (void) sinkID; + (void) notificationConfiguration; + //todo: implement asyncSetSinkNotificationConfiguration; + return (E_NOT_USED); +} + +am_Error_e RoutingSenderPULSE::asyncSetSourceNotificationConfiguration(const am_Handle_s handle, const am_sourceID_t sourceID, const am_NotificationConfiguration_s& notificationConfiguration) +{ + (void) handle; + (void) sourceID; + (void) notificationConfiguration; + //todo: implement asyncSetSourceNotificationConfiguration; + return (E_NOT_USED); +} +/******************************************************************************* + * Private methods + ******************************************************************************/ + + +void RoutingSenderPULSE::getSinkInputInfoCallback(pa_context *c, const pa_sink_input_info *i, void *userdata) +{ + if (i == NULL) + { + return; + } + + //search for corresponding Source + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iter = m_sources.begin(); + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iterEnd = m_sources.end(); + for (; iter < iterEnd; ++iter) + { + //try to match source PulseAudio properties against config properties + const char *property_value = pa_proplist_gets(i->proplist, iter->propertyName.c_str()); + + if (property_value && + ( std::string::npos != std::string(property_value).find(iter->propertyValue) || + std::string::npos != iter->propertyValue.find(property_value)) ) + { + logInfo("PULSE - sink input registered:" + , " sinkInputIndex:", i->index, "sourceID:", iter->source.sourceID); + + logInfo("PULSE - sink input details:" + , " prop_val: ", property_value, " iter->prop_val: ", iter->propertyValue); + + m_sourceToPASinkInput[iter->source.sourceID] = i->index; + + //iterate pending connection request + // -> if there is a connection pending such that sink input "i" matches source from Connect() - create the connection in pulse + std::vector<RoutingSenderPULSEConnection>::iterator iterConn = m_activeConnections.begin(); + std::vector<RoutingSenderPULSEConnection>::iterator iterConnEnd = m_activeConnections.end(); + for (; iterConn < iterConnEnd; ++iterConn) + { + if (iterConn->sourceID == iter->source.sourceID) + { + logInfo("PULSE - asyncConnect() - connectionID:", iterConn->connectionID, + "move sinkInputIndex:", m_sourceToPASinkInput[iterConn->sourceID], "to sinkIndex:", m_sinkToPASink[iterConn->sinkID]); + + routing_sender_move_sink_input( + this->m_paContext, + m_sourceToPASinkInput[iterConn->sourceID], + m_sinkToPASink[iterConn->sinkID], + this); + + //TODO: add callback for pulse move sink input -> to send confirmation; for the moment directly send confirmation + this->m_routingReceiver->ackConnect(iterConn->handle, iterConn->connectionID, am::E_OK); + } + } + //check of controller already requested vol adjustment for this source + bool requiresVolUpdate = false; + for (int j = 0; j < i->volume.channels; j++) + { + if ((i->volume.values[j]*MAX_SOURCE_VOLUME / MAX_PULSE_VOLUME) != m_sourceToVolume[iter->source.sourceID]) + { + requiresVolUpdate = true; + logInfo("PULSE - sink registerd with vol:", (i->volume.values[j]*MAX_SOURCE_VOLUME / MAX_PULSE_VOLUME), + "; should be changed to:", + m_sourceToVolume[iter->source.sourceID]); + break; + } + } + if (requiresVolUpdate) + { + routing_sender_sink_input_volume( + this->m_paContext, + m_sourceToPASinkInput[iter->source.sourceID], + m_sourceToVolume[iter->source.sourceID], + this); + } + //TODO: check mute state was requested by controller. + break; + } + } +} + + +void RoutingSenderPULSE::getSourceOutputInfoCallback(pa_context *c, const pa_source_output_info *i, void *userdata) +{ + if (i == NULL) + { + return; + } + + //search for corresponding Source + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iter = m_sinks.begin(); + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iterEnd = m_sinks.end(); + for (; iter < iterEnd; ++iter) + { + //try to match source PulseAudio properties agains config properties + const char *property_value = pa_proplist_gets(i->proplist, iter->propertyName.c_str()); + + if (property_value && + ( std::string::npos != std::string(property_value).find(iter->propertyValue) || + std::string::npos != iter->propertyValue.find(property_value)) ) + { + logInfo("PULSE - source output registered:" + , " sourceOutputIndex:", i->index, "sinkID:", iter->sink.sinkID); + + m_sinkToPASourceOutput[iter->sink.sinkID] = i->index; + + //iterate pending connection request + // -> if there is a connection pending such that sink input "i" matches source from Connect() - create the connection in pulse + std::vector<RoutingSenderPULSEConnection>::iterator iterConn = m_activeConnections.begin(); + std::vector<RoutingSenderPULSEConnection>::iterator iterConnEnd = m_activeConnections.end(); + for (; iterConn < iterConnEnd; ++iterConn) + { + if (iterConn->sinkID == iter->sink.sinkID) + { + logInfo("PULSE - asyncConnect() - connectionID:", iterConn->connectionID, + "move sourceOutputIndex:", m_sinkToPASourceOutput[iterConn->sinkID], "to sourceIndex:", m_sourceToPASource[iterConn->sourceID]); + + routing_sender_move_source_output( + this->m_paContext, + m_sinkToPASourceOutput[iterConn->sinkID], + m_sourceToPASource[iterConn->sourceID], + this); + + //TODO: add callback for pulse move source output -> to send confirmation; for the moment directly send confirmation + this->m_routingReceiver->ackConnect(iterConn->handle, iterConn->connectionID, am::E_OK); + } + } + + break; + } + } +} + + +void RoutingSenderPULSE::getSinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *userdata) +{ + if (i != NULL) + { + if (strcmp("null", i->name) == 0) + { + this->m_paSinkNullIndex = i->index; + } + + //search for corresponding (already registered) Sink + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iter = m_sinks.begin(); + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iterEnd = m_sinks.end(); + for (; iter < iterEnd; ++iter) + { + //first try to match the sink name from pulse audio sink name + if (iter->sink.name == std::string(i->name)) + { + logInfo("PULSE sink name PA:", i->name, "config name:" ,iter->sink.name); + logInfo("PULSE - PA sink:", i->index, + "corresponding to AMGR sink:", iter->sink.sinkID, " - found"); + m_sinkToPASink[iter->sink.sinkID] = i->index; + } + else + { + //try to match sink PulseAudio properties against config properties + const char *property_value = pa_proplist_gets(i->proplist, iter->propertyName.c_str()); + + if (!property_value) continue; + + if (std::string::npos != iter->propertyValue.find(property_value)) + { + logInfo("PULSE - PA sink:", i->index, + "corresponding to AMGR sink:", iter->sink.sinkID, " - found"); + + m_sinkToPASink[iter->sink.sinkID] = i->index; + } + } + } + } + else if (is_last) + { + routing_sender_get_source_info(this->m_paContext, this); + logInfo("PULSE - PA sinks registration completed"); + } +} + +void RoutingSenderPULSE::getSourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *userdata) +{ + if (i != NULL) + { + if (strcmp("null", i->name) == 0) + { + this->m_paSourceNullIndex = i->index; + } + + //search for corresponding (already registered) Source + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iter = m_sources.begin(); + std::vector<RoutingSenderPULSESourceSinkConfig>::iterator iterEnd = m_sources.end(); + for (; iter < iterEnd; ++iter) + { + //first try to match the sink name from pulse audio sink name + if (iter->sink.name == std::string(i->name)) + { + logInfo("PULSE - PA source:", i->index, + "corresponding to AMGR source:", iter->source.sourceID, " - found"); + m_sourceToPASource[iter->source.sourceID] = i->index; + } + else + { + //try to match source PulseAudio properties against config properties + const char *property_value = pa_proplist_gets(i->proplist, iter->propertyName.c_str()); + + if (!property_value) continue; + + if (std::string::npos != iter->propertyValue.find(property_value)) + { + logInfo("PULSE - PA source:", i->index, + "corresponding to AMGR source:", iter->source.sourceID, " - found"); + + m_sourceToPASource[iter->source.sourceID] = i->index; + } + } + + } + } + else if (is_last) + { + this->m_routingReceiver->hookDomainRegistrationComplete(this->m_domain.domainID); + logInfo("PULSE - PA sinks and source registration completed"); + //TODO: - search for existing sink inputs & sources outputs + } +} |