526 lines
16 KiB
C
526 lines
16 KiB
C
/*
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* Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
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* Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
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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, MA 02111-1307 USA
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*/
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#ifndef __PJMEDIA_AUDIODEV_CONFIG_H__
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#define __PJMEDIA_AUDIODEV_CONFIG_H__
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/**
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* @file config.h
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* @brief Audio config.
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*/
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#include <pjmedia/types.h>
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#include <pj/pool.h>
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PJ_BEGIN_DECL
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/**
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* @defgroup audio_device_api Audio Device API
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* @brief PJMEDIA audio device abstraction API.
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*/
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/**
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* @defgroup s1_audio_device_config Compile time configurations
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* @ingroup audio_device_api
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* @brief Compile time configurations
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* @{
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*/
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/**
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* This setting controls the buffer length of audio device name.
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*
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* Default: 128 for Windows platforms, 64 for others
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*/
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#ifndef PJMEDIA_AUD_DEV_INFO_NAME_LEN
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# if (defined(PJ_WIN32) && PJ_WIN32!=0) || \
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(defined(PJ_WIN64) && PJ_WIN64!=0)
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# define PJMEDIA_AUD_DEV_INFO_NAME_LEN 128
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# else
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# define PJMEDIA_AUD_DEV_INFO_NAME_LEN 64
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# endif
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#endif
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/**
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* This setting controls whether PortAudio support should be included.
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*
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* By default it is disabled.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO
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# define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0
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#endif
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/**
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* This setting controls whether Android OpenSL audio support should be
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* included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_OPENSL
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# define PJMEDIA_AUDIO_DEV_HAS_OPENSL 0
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#endif
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/**
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* This setting controls whether Android JNI audio support should be
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* included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_ANDROID_JNI
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# define PJMEDIA_AUDIO_DEV_HAS_ANDROID_JNI PJ_ANDROID
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#endif
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/**
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* This setting controls whether Android Oboe audio support should be
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* included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_OBOE
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# define PJMEDIA_AUDIO_DEV_HAS_OBOE 0
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#endif
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/**
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* This setting controls whether BlackBerry 10 (BB10) audio support
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* should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_BB10
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# define PJMEDIA_AUDIO_DEV_HAS_BB10 0
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#endif
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/**
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* This setting controls whether native ALSA support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_ALSA
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# define PJMEDIA_AUDIO_DEV_HAS_ALSA 0
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#endif
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/**
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* This setting controls whether null audio support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_NULL_AUDIO
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# define PJMEDIA_AUDIO_DEV_HAS_NULL_AUDIO 0
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#endif
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/**
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* This setting controls whether coreaudio support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_COREAUDIO
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# define PJMEDIA_AUDIO_DEV_HAS_COREAUDIO 0
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#endif
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/**
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* This setting controls whether WMME support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_WMME
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# if (defined(PJ_WIN32_UWP) && PJ_WIN32_UWP!=0) || \
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(defined(PJ_WIN32_WINPHONE8) && PJ_WIN32_WINPHONE8!=0)
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# define PJMEDIA_AUDIO_DEV_HAS_WMME 0
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# else
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# define PJMEDIA_AUDIO_DEV_HAS_WMME 1
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# endif
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#endif
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/**
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* This setting controls whether Windows Audio Session API (WASAPI)
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* support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_WASAPI
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# if (defined(PJ_WIN32_UWP) && PJ_WIN32_UWP!=0) || \
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(defined(PJ_WIN32_WINPHONE8) && PJ_WIN32_WINPHONE8!=0)
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# define PJMEDIA_AUDIO_DEV_HAS_WASAPI 1
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# else
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# define PJMEDIA_AUDIO_DEV_HAS_WASAPI 0
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# endif
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#endif
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/**
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* This setting controls whether BDIMAD support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_BDIMAD
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# define PJMEDIA_AUDIO_DEV_HAS_BDIMAD 0
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#endif
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/**
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* This setting controls whether Symbian APS support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_SYMB_APS
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# define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 0
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#endif
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/**
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* This setting controls whether Symbian APS should perform codec
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* detection in its factory initalization. Note that codec detection
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* may take few seconds and detecting more codecs will take more time.
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* Possible values are:
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* - 0: no codec detection, all APS codec (AMR-NB, G.711, G.729, and
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* iLBC) will be assumed as supported.
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* - 1: minimal codec detection, i.e: only detect for AMR-NB and G.711,
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* (G.729 and iLBC are considered to be supported/unsupported when
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* G.711 is supported/unsupported).
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* - 2: full codec detection, i.e: detect AMR-NB, G.711, G.729, and iLBC.
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*
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* Default: 1 (minimal codec detection)
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*/
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#ifndef PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC
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# define PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC 1
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#endif
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/**
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* This setting controls whether Symbian VAS support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS
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# define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0
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#endif
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/**
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* This setting controls Symbian VAS version to be used. Currently, valid
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* values are only 1 (for VAS 1.0) and 2 (for VAS 2.0).
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*
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* Default: 1 (VAS version 1.0)
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*/
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#ifndef PJMEDIA_AUDIO_DEV_SYMB_VAS_VERSION
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# define PJMEDIA_AUDIO_DEV_SYMB_VAS_VERSION 1
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#endif
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/**
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* This setting controls whether Symbian audio (using built-in multimedia
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* framework) support should be included.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA
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# define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA PJ_SYMBIAN
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#endif
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/**
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* This setting controls whether the Symbian audio with built-in multimedia
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* framework backend should be started synchronously. Note that synchronous
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* start will block the application/UI, e.g: about 40ms for each direction
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* on N95. While asynchronous start may cause invalid value (always zero)
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* returned in input/output volume query, if the query is performed when
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* the internal start procedure is not completely finished.
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*
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* Default: 1 (yes)
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*/
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#ifndef PJMEDIA_AUDIO_DEV_MDA_USE_SYNC_START
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# define PJMEDIA_AUDIO_DEV_MDA_USE_SYNC_START 1
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#endif
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/**
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* This setting controls whether the Audio Device API should support
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* device implementation that is based on the old sound device API
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* (sound.h).
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*
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* Enable this API if:
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* - you have implemented your own sound device using the old sound
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* device API (sound.h), and
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* - you wish to be able to use your sound device implementation
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* using the new Audio Device API.
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*
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* Please see http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more
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* info.
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*/
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#ifndef PJMEDIA_AUDIO_DEV_HAS_LEGACY_DEVICE
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# define PJMEDIA_AUDIO_DEV_HAS_LEGACY_DEVICE 0
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#endif
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/**
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* @}
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*/
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PJ_END_DECL
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#endif /* __PJMEDIA_AUDIODEV_CONFIG_H__ */
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/*
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--------------------- DOCUMENTATION FOLLOWS ---------------------------
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*/
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/**
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* @addtogroup audio_device_api Audio Device API
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* @{
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PJMEDIA Audio Device API is a cross-platform audio API appropriate for use with
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VoIP applications and many other types of audio streaming applications.
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The API abstracts many different audio API's on various platforms, such as:
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- WMME audio for Windows and Windows Mobile devices
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- Windows Audio Session API (WASAPI)
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- CoreAudio for Mac and iPhone
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- ALSA for Linux
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- Android OpenSL
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- Android JNI
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- Android Oboe
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- PortAudio for Windows, Windows Mobile, Linux, Unix, dan MacOS X.
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- BDIMAP
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- Blackberry BB10
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- Symbian audio streaming/multimedia framework (MMF) implementation
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- Nokia Audio Proxy Server (APS) implementation
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- null-audio implementation
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- and more to be implemented in the future
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The Audio Device API/library is an evolution from PJMEDIA @ref PJMED_SND and
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contains many enhancements:
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- Forward compatibility:
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\n
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The new API has been designed to be extensible, it will support new API's as
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well as new features that may be introduced in the future without breaking
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compatibility with applications that use this API as well as compatibility
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with existing device implementations.
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- Device capabilities:
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\n
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At the heart of the API is device capabilities management, where all possible
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audio capabilities of audio devices should be able to be handled in a generic
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manner. With this framework, new capabilities that may be discovered in the
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future can be handled in manner without breaking existing applications.
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- Built-in features:
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\n
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The device capabilities framework enables applications to use and control
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audio features built-in in the device, such as:
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- echo cancellation,
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- built-in codecs,
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- audio routing (e.g. to earpiece or loudspeaker),
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- volume control,
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- etc.
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- Codec support:
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\n
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Some audio devices such as Nokia/Symbian Audio Proxy Server (APS) and Nokia
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VoIP Audio Services (VAS) support built-in hardware audio codecs (e.g. G.729,
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iLBC, and AMR), and application can use the sound device in encoded mode to
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make use of these hardware codecs.
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- Multiple backends:
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\n
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The new API supports multiple audio backends (called factories or drivers in
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the code) to be active simultaneously, and audio backends may be added or
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removed during run-time.
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@section using Overview on using the API
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@subsection getting_started Getting started
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-# <b>Configure the application's project settings</b>.\n
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Add the following
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include:
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\code
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#include <pjmedia_audiodev.h>\endcode\n
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And add <b>pjmedia-audiodev</b> library to your application link
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specifications.\n
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-# <b>Compile time settings</b>.\n
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Use the compile time settings to enable or
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disable specific audio drivers. For more information, please see
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\ref s1_audio_device_config.
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-# <b>API initialization and cleaning up</b>.\n
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Before anything else, application must initialize the API by calling:
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\code
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pjmedia_aud_subsys_init(pf);\endcode\n
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And add this in the application cleanup sequence
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\code
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pjmedia_aud_subsys_shutdown();\endcode
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@subsection devices Working with devices
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-# The following code prints the list of audio devices detected
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in the system.
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\code
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int dev_count;
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pjmedia_aud_dev_index dev_idx;
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pj_status_t status;
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dev_count = pjmedia_aud_dev_count();
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printf("Got %d audio devices\n", dev_count);
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for (dev_idx=0; dev_idx<dev_count; ++i) {
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pjmedia_aud_dev_info info;
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status = pjmedia_aud_dev_get_info(dev_idx, &info);
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printf("%d. %s (in=%d, out=%d)\n",
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dev_idx, info.name,
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info.input_count, info.output_count);
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}
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\endcode\n
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-# Info: The #PJMEDIA_AUD_DEFAULT_CAPTURE_DEV and #PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV
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constants are used to denote default capture and playback devices
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respectively.
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-# Info: You may save the device and driver's name in your application
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setting, for example to specify the prefered devices to be
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used by your application. You can then retrieve the device index
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for the device by calling:
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\code
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const char *drv_name = "WMME";
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const char *dev_name = "Wave mapper";
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pjmedia_aud_dev_index dev_idx;
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status = pjmedia_aud_dev_lookup(drv_name, dev_name, &dev_idx);
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if (status==PJ_SUCCESS)
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printf("Device index is %d\n", dev_idx);
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\endcode
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@subsection caps Device capabilities
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Capabilities are encoded as #pjmedia_aud_dev_cap enumeration. Please see
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#pjmedia_aud_dev_cap enumeration for more information.
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-# The following snippet prints the capabilities supported by the device:
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\code
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pjmedia_aud_dev_info info;
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pj_status_t status;
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status = pjmedia_aud_dev_get_info(PJMEDIA_AUD_DEFAULT_CAPTURE_DEV, &info);
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if (status == PJ_SUCCESS) {
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unsigned i;
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// Enumerate capability bits
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printf("Device capabilities: ");
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for (i=0; i<32; ++i) {
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if (info.caps & (1 << i))
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printf("%s ", pjmedia_aud_dev_cap_name(1 << i, NULL));
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}
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}
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\endcode\n
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-# Info: You can set the device settings when opening audio stream by setting
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the flags and the appropriate setting in #pjmedia_aud_param when calling
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#pjmedia_aud_stream_create()\n
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-# Info: Once the audio stream is running, you can retrieve or change the stream
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setting by specifying the capability in #pjmedia_aud_stream_get_cap()
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and #pjmedia_aud_stream_set_cap() respectively.
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@subsection creating_stream Creating audio streams
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The audio stream enables audio streaming to capture device, playback device,
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or both.
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-# It is recommended to initialize the #pjmedia_aud_param with its default
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values before using it:
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\code
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pjmedia_aud_param param;
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pjmedia_aud_dev_index dev_idx;
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pj_status_t status;
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dev_idx = PJMEDIA_AUD_DEFAULT_CAPTURE_DEV;
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status = pjmedia_aud_dev_default_param(dev_idx, ¶m);
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\endcode\n
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-# Configure the mandatory parameters:
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\code
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param.dir = PJMEDIA_DIR_CAPTURE_PLAYBACK;
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param.rec_id = PJMEDIA_AUD_DEFAULT_CAPTURE_DEV;
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param.play_id = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV;
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param.clock_rate = 8000;
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param.channel_count = 1;
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param.samples_per_frame = 160;
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param.bits_per_sample = 16;
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\endcode\n
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-# If you want the audio stream to use the device's built-in codec, specify
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the codec in the #pjmedia_aud_param. You must make sure that the codec
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is supported by the device, by looking at its supported format list in
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the #pjmedia_aud_dev_info.\n
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The snippet below sets the audio stream to use G.711 ULAW encoding:
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\code
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unsigned i;
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// Make sure Ulaw is supported
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if ((info.caps & PJMEDIA_AUD_DEV_CAP_EXT_FORMAT) == 0)
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error("Device does not support extended formats");
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for (i = 0; i < info.ext_fmt_cnt; ++i) {
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if (info.ext_fmt[i].id == PJMEDIA_FORMAT_ULAW)
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break;
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}
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if (i == info.ext_fmt_cnt)
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error("Device does not support Ulaw format");
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// Set Ulaw format
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param.flags |= PJMEDIA_AUD_DEV_CAP_EXT_FORMAT;
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param.ext_fmt.id = PJMEDIA_FORMAT_ULAW;
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param.ext_fmt.bitrate = 64000;
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param.ext_fmt.vad = PJ_FALSE;
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\endcode\n
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-# Note that if non-PCM format is configured on the audio stream, the
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capture and/or playback functions (#pjmedia_aud_rec_cb and
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#pjmedia_aud_play_cb respectively) will report the audio frame as
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#pjmedia_frame_ext structure instead of the #pjmedia_frame.
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-# Optionally configure other device's capabilities. The following snippet
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shows how to enable echo cancellation on the device (note that this
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snippet may not be necessary since the setting may have been enabled
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when calling #pjmedia_aud_dev_default_param() above):
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\code
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if (info.caps & PJMEDIA_AUD_DEV_CAP_EC) {
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param.flags |= PJMEDIA_AUD_DEV_CAP_EC;
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param.ec_enabled = PJ_TRUE;
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}
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\endcode
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-# Open the audio stream, specifying the capture and/or playback callback
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functions:
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\code
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pjmedia_aud_stream *stream;
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status = pjmedia_aud_stream_create(¶m, &rec_cb, &play_cb,
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user_data, &stream);
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\endcode
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@subsection working_with_stream Working with audio streams
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-# To start the audio stream:
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\code
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status = pjmedia_aud_stream_start(stream);
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\endcode\n
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To stop the stream:
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\code
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status = pjmedia_aud_stream_stop(stream);
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\endcode\n
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And to destroy the stream:
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\code
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status = pjmedia_aud_stream_destroy(stream);
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\endcode\n
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-# Info: The following shows how to retrieve the capability value of the
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stream (in this case, the current output volume setting).
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\code
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// Volume setting is an unsigned integer showing the level in percent.
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unsigned vol;
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status = pjmedia_aud_stream_get_cap(stream,
|
|
PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING,
|
|
&vol);
|
|
\endcode
|
|
-# Info: And following shows how to modify the capability value of the
|
|
stream (in this case, the current output volume setting).
|
|
\code
|
|
// Volume setting is an unsigned integer showing the level in percent.
|
|
unsigned vol = 50;
|
|
status = pjmedia_aud_stream_set_cap(stream,
|
|
PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING,
|
|
&vol);
|
|
\endcode
|
|
|
|
|
|
*/
|
|
|
|
|
|
/**
|
|
* @}
|
|
*/
|
|
|