diff options
Diffstat (limited to 'pjsip-apps/src/3rdparty_media_sample/alt_pjsua_aud.c')
| -rw-r--r-- | pjsip-apps/src/3rdparty_media_sample/alt_pjsua_aud.c | 631 |
1 files changed, 631 insertions, 0 deletions
diff --git a/pjsip-apps/src/3rdparty_media_sample/alt_pjsua_aud.c b/pjsip-apps/src/3rdparty_media_sample/alt_pjsua_aud.c new file mode 100644 index 0000000..fd2bf71 --- /dev/null +++ b/pjsip-apps/src/3rdparty_media_sample/alt_pjsua_aud.c @@ -0,0 +1,631 @@ +/* $Id: alt_pjsua_aud.c 4174 2012-06-21 08:09:53Z bennylp $ */ +/* + * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <pjsua-lib/pjsua.h> +#include <pjsua-lib/pjsua_internal.h> + +#if defined(PJSUA_MEDIA_HAS_PJMEDIA) && PJSUA_MEDIA_HAS_PJMEDIA != 0 +# error The PJSUA_MEDIA_HAS_PJMEDIA should be declared as zero +#endif + + +#define THIS_FILE "alt_pjsua_aud.c" +#define UNIMPLEMENTED(func) PJ_LOG(2,(THIS_FILE, "*** Call to unimplemented function %s ***", #func)); + + +/***************************************************************************** + * Our dummy codecs. Since we won't use any PJMEDIA codecs, we need to declare + * our own codecs and register them to PJMEDIA's codec manager. We just need + * the info so that they can be listed in SDP. The encoding and decoding will + * happen in your third party media stream and will not use these codecs, + * hence the "dummy" name. + */ +static struct alt_codec +{ + pj_str_t encoding_name; + pj_uint8_t payload_type; + unsigned clock_rate; + unsigned channel_cnt; + unsigned frm_ptime; + unsigned avg_bps; + unsigned max_bps; +} codec_list[] = +{ + /* G.729 */ + { { "G729", 4 }, 18, 8000, 1, 10, 8000, 8000 }, + /* PCMU */ + { { "PCMU", 4 }, 0, 8000, 1, 10, 64000, 64000 }, + /* Our proprietary high end low bit rate (5kbps) codec, if you wish */ + { { "FOO", 3 }, PJMEDIA_RTP_PT_START+0, 16000, 1, 20, 5000, 5000 }, +}; + +static struct alt_codec_factory +{ + pjmedia_codec_factory base; +} alt_codec_factory; + +static pj_status_t alt_codec_test_alloc( pjmedia_codec_factory *factory, + const pjmedia_codec_info *id ) +{ + unsigned i; + for (i=0; i<PJ_ARRAY_SIZE(codec_list); ++i) { + if (pj_stricmp(&id->encoding_name, &codec_list[i].encoding_name)==0) + return PJ_SUCCESS; + } + return PJ_ENOTSUP; +} + +static pj_status_t alt_codec_default_attr( pjmedia_codec_factory *factory, + const pjmedia_codec_info *id, + pjmedia_codec_param *attr ) +{ + struct alt_codec *ac; + unsigned i; + + PJ_UNUSED_ARG(factory); + + for (i=0; i<PJ_ARRAY_SIZE(codec_list); ++i) { + if (pj_stricmp(&id->encoding_name, &codec_list[i].encoding_name)==0) + break; + } + if (i == PJ_ARRAY_SIZE(codec_list)) + return PJ_ENOTFOUND; + + ac = &codec_list[i]; + + pj_bzero(attr, sizeof(pjmedia_codec_param)); + attr->info.clock_rate = ac->clock_rate; + attr->info.channel_cnt = ac->channel_cnt; + attr->info.avg_bps = ac->avg_bps; + attr->info.max_bps = ac->max_bps; + attr->info.pcm_bits_per_sample = 16; + attr->info.frm_ptime = ac->frm_ptime; + attr->info.pt = ac->payload_type; + + attr->setting.frm_per_pkt = 1; + attr->setting.vad = 1; + attr->setting.plc = 1; + + return PJ_SUCCESS; +} + +static pj_status_t alt_codec_enum_codecs(pjmedia_codec_factory *factory, + unsigned *count, + pjmedia_codec_info codecs[]) +{ + unsigned i; + + for (i=0; i<*count && i<PJ_ARRAY_SIZE(codec_list); ++i) { + struct alt_codec *ac = &codec_list[i]; + pj_bzero(&codecs[i], sizeof(pjmedia_codec_info)); + codecs[i].encoding_name = ac->encoding_name; + codecs[i].pt = ac->payload_type; + codecs[i].type = PJMEDIA_TYPE_AUDIO; + codecs[i].clock_rate = ac->clock_rate; + codecs[i].channel_cnt = ac->channel_cnt; + } + + *count = i; + + return PJ_SUCCESS; +} + +static pj_status_t alt_codec_alloc_codec(pjmedia_codec_factory *factory, + const pjmedia_codec_info *id, + pjmedia_codec **p_codec) +{ + /* This will never get called since we won't be using this codec */ + UNIMPLEMENTED(alt_codec_alloc_codec) + return PJ_ENOTSUP; +} + +static pj_status_t alt_codec_dealloc_codec( pjmedia_codec_factory *factory, + pjmedia_codec *codec ) +{ + /* This will never get called */ + UNIMPLEMENTED(alt_codec_dealloc_codec) + return PJ_ENOTSUP; +} + +static pj_status_t alt_codec_deinit(void) +{ + pjmedia_codec_mgr *codec_mgr; + codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt); + return pjmedia_codec_mgr_unregister_factory(codec_mgr, + &alt_codec_factory.base); + +} + +static pjmedia_codec_factory_op alt_codec_factory_op = +{ + &alt_codec_test_alloc, + &alt_codec_default_attr, + &alt_codec_enum_codecs, + &alt_codec_alloc_codec, + &alt_codec_dealloc_codec, + &alt_codec_deinit +}; + + +/***************************************************************************** + * API + */ + +/* Initialize third party media library. */ +pj_status_t pjsua_aud_subsys_init() +{ + pjmedia_codec_mgr *codec_mgr; + pj_status_t status; + + /* Register our "dummy" codecs */ + alt_codec_factory.base.op = &alt_codec_factory_op; + codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt); + status = pjmedia_codec_mgr_register_factory(codec_mgr, + &alt_codec_factory.base); + if (status != PJ_SUCCESS) + return status; + + /* TODO: initialize your evil library here */ + return PJ_SUCCESS; +} + +/* Start (audio) media library. */ +pj_status_t pjsua_aud_subsys_start(void) +{ + /* TODO: */ + return PJ_SUCCESS; +} + +/* Cleanup and deinitialize third party media library. */ +pj_status_t pjsua_aud_subsys_destroy() +{ + /* TODO: */ + return PJ_SUCCESS; +} + +/* Our callback to receive incoming RTP packets */ +static void aud_rtp_cb(void *user_data, void *pkt, pj_ssize_t size) +{ + pjsua_call_media *call_med = (pjsua_call_media*) user_data; + + /* TODO: Do something with the packet */ + PJ_LOG(4,(THIS_FILE, "RX %d bytes audio RTP packet", (int)size)); +} + +/* Our callback to receive RTCP packets */ +static void aud_rtcp_cb(void *user_data, void *pkt, pj_ssize_t size) +{ + pjsua_call_media *call_med = (pjsua_call_media*) user_data; + + /* TODO: Do something with the packet here */ + PJ_LOG(4,(THIS_FILE, "RX %d bytes audio RTCP packet", (int)size)); +} + +/* A demo function to send dummy "RTP" packets periodically. You would not + * need to have this function in the real app! + */ +static void timer_to_send_aud_rtp(void *user_data) +{ + pjsua_call_media *call_med = (pjsua_call_media*) user_data; + const char *pkt = "Not RTP packet"; + + if (!call_med->call || !call_med->call->inv || !call_med->tp) { + /* Call has been disconnected. There is race condition here as + * this cb may be called sometime after call has been disconnected */ + return; + } + + pjmedia_transport_send_rtp(call_med->tp, pkt, strlen(pkt)); + + pjsua_schedule_timer2(&timer_to_send_aud_rtp, call_med, 2000); +} + +static void timer_to_send_aud_rtcp(void *user_data) +{ + pjsua_call_media *call_med = (pjsua_call_media*) user_data; + const char *pkt = "Not RTCP packet"; + + if (!call_med->call || !call_med->call->inv || !call_med->tp) { + /* Call has been disconnected. There is race condition here as + * this cb may be called sometime after call has been disconnected */ + return; + } + + pjmedia_transport_send_rtcp(call_med->tp, pkt, strlen(pkt)); + + pjsua_schedule_timer2(&timer_to_send_aud_rtcp, call_med, 5000); +} + +/* Stop the audio stream of a call. */ +void pjsua_aud_stop_stream(pjsua_call_media *call_med) +{ + /* Detach our RTP/RTCP callbacks from transport */ + pjmedia_transport_detach(call_med->tp, call_med); + + /* TODO: destroy your audio stream here */ +} + +/* + * This function is called whenever SDP negotiation has completed + * successfully. Here you'd want to start your audio stream + * based on the info in the SDPs. + */ +pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med, + pj_pool_t *tmp_pool, + pjmedia_stream_info *si, + const pjmedia_sdp_session *local_sdp, + const pjmedia_sdp_session *remote_sdp) +{ + pj_status_t status = PJ_SUCCESS; + + PJ_LOG(4,(THIS_FILE,"Alt audio channel update..")); + pj_log_push_indent(); + + /* Check if no media is active */ + if (si->dir != PJMEDIA_DIR_NONE) { + /* Attach our RTP and RTCP callbacks to the media transport */ + status = pjmedia_transport_attach(call_med->tp, call_med, + &si->rem_addr, &si->rem_rtcp, + pj_sockaddr_get_len(&si->rem_addr), + &aud_rtp_cb, &aud_rtcp_cb); + + /* For a demonstration, let's use a timer to send "RTP" packet + * periodically. + */ + pjsua_schedule_timer2(&timer_to_send_aud_rtp, call_med, 0); + pjsua_schedule_timer2(&timer_to_send_aud_rtcp, call_med, 2500); + + /* TODO: + * - Create and start your media stream based on the parameters + * in si + */ + } + +on_return: + pj_log_pop_indent(); + return status; +} + +void pjsua_check_snd_dev_idle() +{ +} + +/***************************************************************************** + * + * Call API which MAY need to be re-implemented if different backend is used. + */ + +/* Check if call has an active media session. */ +PJ_DEF(pj_bool_t) pjsua_call_has_media(pjsua_call_id call_id) +{ + UNIMPLEMENTED(pjsua_call_has_media) + return PJ_TRUE; +} + + +/* Get the conference port identification associated with the call. */ +PJ_DEF(pjsua_conf_port_id) pjsua_call_get_conf_port(pjsua_call_id call_id) +{ + UNIMPLEMENTED(pjsua_call_get_conf_port) + return PJSUA_INVALID_ID; +} + +/* Get media stream info for the specified media index. */ +PJ_DEF(pj_status_t) pjsua_call_get_stream_info( pjsua_call_id call_id, + unsigned med_idx, + pjsua_stream_info *psi) +{ + pj_bzero(psi, sizeof(*psi)); + UNIMPLEMENTED(pjsua_call_get_stream_info) + return PJ_ENOTSUP; +} + +/* Get media stream statistic for the specified media index. */ +PJ_DEF(pj_status_t) pjsua_call_get_stream_stat( pjsua_call_id call_id, + unsigned med_idx, + pjsua_stream_stat *stat) +{ + pj_bzero(stat, sizeof(*stat)); + UNIMPLEMENTED(pjsua_call_get_stream_stat) + return PJ_ENOTSUP; +} + +/* + * Send DTMF digits to remote using RFC 2833 payload formats. + */ +PJ_DEF(pj_status_t) pjsua_call_dial_dtmf( pjsua_call_id call_id, + const pj_str_t *digits) +{ + UNIMPLEMENTED(pjsua_call_dial_dtmf) + return PJ_ENOTSUP; +} + +/***************************************************************************** + * Below are auxiliary API that we don't support (feel free to implement them + * with the other media stack) + */ + +/* Get maximum number of conference ports. */ +PJ_DEF(unsigned) pjsua_conf_get_max_ports(void) +{ + UNIMPLEMENTED(pjsua_conf_get_max_ports) + return 0xFF; +} + +/* Get current number of active ports in the bridge. */ +PJ_DEF(unsigned) pjsua_conf_get_active_ports(void) +{ + UNIMPLEMENTED(pjsua_conf_get_active_ports) + return 0; +} + +/* Enumerate all conference ports. */ +PJ_DEF(pj_status_t) pjsua_enum_conf_ports(pjsua_conf_port_id id[], + unsigned *count) +{ + *count = 0; + UNIMPLEMENTED(pjsua_enum_conf_ports) + return PJ_ENOTSUP; +} + +/* Get information about the specified conference port */ +PJ_DEF(pj_status_t) pjsua_conf_get_port_info( pjsua_conf_port_id id, + pjsua_conf_port_info *info) +{ + UNIMPLEMENTED(pjsua_conf_get_port_info) + return PJ_ENOTSUP; +} + +/* Add arbitrary media port to PJSUA's conference bridge. */ +PJ_DEF(pj_status_t) pjsua_conf_add_port( pj_pool_t *pool, + pjmedia_port *port, + pjsua_conf_port_id *p_id) +{ + *p_id = PJSUA_INVALID_ID; + UNIMPLEMENTED(pjsua_conf_add_port) + /* We should return PJ_ENOTSUP here, but this API is needed by pjsua + * application or otherwise it will refuse to start. + */ + return PJ_SUCCESS; +} + +/* Remove arbitrary slot from the conference bridge. */ +PJ_DEF(pj_status_t) pjsua_conf_remove_port(pjsua_conf_port_id id) +{ + UNIMPLEMENTED(pjsua_conf_remove_port) + return PJ_ENOTSUP; +} + +/* Establish unidirectional media flow from souce to sink. */ +PJ_DEF(pj_status_t) pjsua_conf_connect( pjsua_conf_port_id source, + pjsua_conf_port_id sink) +{ + UNIMPLEMENTED(pjsua_conf_connect) + return PJ_ENOTSUP; +} + +/* Disconnect media flow from the source to destination port. */ +PJ_DEF(pj_status_t) pjsua_conf_disconnect( pjsua_conf_port_id source, + pjsua_conf_port_id sink) +{ + UNIMPLEMENTED(pjsua_conf_disconnect) + return PJ_ENOTSUP; +} + +/* Adjust the signal level to be transmitted from the bridge to the + * specified port by making it louder or quieter. + */ +PJ_DEF(pj_status_t) pjsua_conf_adjust_tx_level(pjsua_conf_port_id slot, + float level) +{ + UNIMPLEMENTED(pjsua_conf_adjust_tx_level) + return PJ_ENOTSUP; +} + +/* Adjust the signal level to be received from the specified port (to + * the bridge) by making it louder or quieter. + */ +PJ_DEF(pj_status_t) pjsua_conf_adjust_rx_level(pjsua_conf_port_id slot, + float level) +{ + UNIMPLEMENTED(pjsua_conf_adjust_rx_level) + return PJ_ENOTSUP; +} + + +/* Get last signal level transmitted to or received from the specified port. */ +PJ_DEF(pj_status_t) pjsua_conf_get_signal_level(pjsua_conf_port_id slot, + unsigned *tx_level, + unsigned *rx_level) +{ + UNIMPLEMENTED(pjsua_conf_get_signal_level) + return PJ_ENOTSUP; +} + +/* Create a file player, and automatically connect this player to + * the conference bridge. + */ +PJ_DEF(pj_status_t) pjsua_player_create( const pj_str_t *filename, + unsigned options, + pjsua_player_id *p_id) +{ + UNIMPLEMENTED(pjsua_player_create) + return PJ_ENOTSUP; +} + +/* Create a file playlist media port, and automatically add the port + * to the conference bridge. + */ +PJ_DEF(pj_status_t) pjsua_playlist_create( const pj_str_t file_names[], + unsigned file_count, + const pj_str_t *label, + unsigned options, + pjsua_player_id *p_id) +{ + UNIMPLEMENTED(pjsua_playlist_create) + return PJ_ENOTSUP; +} + +/* Get conference port ID associated with player. */ +PJ_DEF(pjsua_conf_port_id) pjsua_player_get_conf_port(pjsua_player_id id) +{ + UNIMPLEMENTED(pjsua_player_get_conf_port) + return -1; +} + +/* Get the media port for the player. */ +PJ_DEF(pj_status_t) pjsua_player_get_port( pjsua_player_id id, + pjmedia_port **p_port) +{ + UNIMPLEMENTED(pjsua_player_get_port) + return PJ_ENOTSUP; +} + +/* Set playback position. */ +PJ_DEF(pj_status_t) pjsua_player_set_pos( pjsua_player_id id, + pj_uint32_t samples) +{ + UNIMPLEMENTED(pjsua_player_set_pos) + return PJ_ENOTSUP; +} + +/* Close the file, remove the player from the bridge, and free + * resources associated with the file player. + */ +PJ_DEF(pj_status_t) pjsua_player_destroy(pjsua_player_id id) +{ + UNIMPLEMENTED(pjsua_player_destroy) + return PJ_ENOTSUP; +} + +/* Create a file recorder, and automatically connect this recorder to + * the conference bridge. + */ +PJ_DEF(pj_status_t) pjsua_recorder_create( const pj_str_t *filename, + unsigned enc_type, + void *enc_param, + pj_ssize_t max_size, + unsigned options, + pjsua_recorder_id *p_id) +{ + UNIMPLEMENTED(pjsua_recorder_create) + return PJ_ENOTSUP; +} + + +/* Get conference port associated with recorder. */ +PJ_DEF(pjsua_conf_port_id) pjsua_recorder_get_conf_port(pjsua_recorder_id id) +{ + UNIMPLEMENTED(pjsua_recorder_get_conf_port) + return -1; +} + +/* Get the media port for the recorder. */ +PJ_DEF(pj_status_t) pjsua_recorder_get_port( pjsua_recorder_id id, + pjmedia_port **p_port) +{ + UNIMPLEMENTED(pjsua_recorder_get_port) + return PJ_ENOTSUP; +} + +/* Destroy recorder (this will complete recording). */ +PJ_DEF(pj_status_t) pjsua_recorder_destroy(pjsua_recorder_id id) +{ + UNIMPLEMENTED(pjsua_recorder_destroy) + return PJ_ENOTSUP; +} + +/* Enum sound devices. */ +PJ_DEF(pj_status_t) pjsua_enum_aud_devs( pjmedia_aud_dev_info info[], + unsigned *count) +{ + UNIMPLEMENTED(pjsua_enum_aud_devs) + return PJ_ENOTSUP; +} + +PJ_DEF(pj_status_t) pjsua_enum_snd_devs( pjmedia_snd_dev_info info[], + unsigned *count) +{ + UNIMPLEMENTED(pjsua_enum_snd_devs) + return PJ_ENOTSUP; +} + +/* Select or change sound device. */ +PJ_DEF(pj_status_t) pjsua_set_snd_dev( int capture_dev, int playback_dev) +{ + UNIMPLEMENTED(pjsua_set_snd_dev) + return PJ_SUCCESS; +} + +/* Get currently active sound devices. */ +PJ_DEF(pj_status_t) pjsua_get_snd_dev(int *capture_dev, int *playback_dev) +{ + *capture_dev = *playback_dev = PJSUA_INVALID_ID; + UNIMPLEMENTED(pjsua_get_snd_dev) + return PJ_ENOTSUP; +} + +/* Use null sound device. */ +PJ_DEF(pj_status_t) pjsua_set_null_snd_dev(void) +{ + UNIMPLEMENTED(pjsua_set_null_snd_dev) + return PJ_ENOTSUP; +} + +/* Use no device! */ +PJ_DEF(pjmedia_port*) pjsua_set_no_snd_dev(void) +{ + UNIMPLEMENTED(pjsua_set_no_snd_dev) + return NULL; +} + +/* Configure the AEC settings of the sound port. */ +PJ_DEF(pj_status_t) pjsua_set_ec(unsigned tail_ms, unsigned options) +{ + UNIMPLEMENTED(pjsua_set_ec) + return PJ_ENOTSUP; +} + +/* Get current AEC tail length. */ +PJ_DEF(pj_status_t) pjsua_get_ec_tail(unsigned *p_tail_ms) +{ + UNIMPLEMENTED(pjsua_get_ec_tail) + return PJ_ENOTSUP; +} + +/* Check whether the sound device is currently active. */ +PJ_DEF(pj_bool_t) pjsua_snd_is_active(void) +{ + UNIMPLEMENTED(pjsua_snd_is_active) + return PJ_FALSE; +} + +/* Configure sound device setting to the sound device being used. */ +PJ_DEF(pj_status_t) pjsua_snd_set_setting( pjmedia_aud_dev_cap cap, + const void *pval, pj_bool_t keep) +{ + UNIMPLEMENTED(pjsua_snd_set_setting) + return PJ_ENOTSUP; +} + +/* Retrieve a sound device setting. */ +PJ_DEF(pj_status_t) pjsua_snd_get_setting(pjmedia_aud_dev_cap cap, void *pval) +{ + UNIMPLEMENTED(pjsua_snd_get_setting) + return PJ_ENOTSUP; +} |