* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/.svn/pristine/de/de076ac5c515836faccb782beb3068e1f65e5de3.svn-base b/jni/pjproject-android/.svn/pristine/de/de076ac5c515836faccb782beb3068e1f65e5de3.svn-base
new file mode 100644
index 0000000..2b85421
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/de/de076ac5c515836faccb782beb3068e1f65e5de3.svn-base
@@ -0,0 +1,631 @@
+/* $Id$ */
+/*
+ * 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;
+}