Started audio bypass tester

mediastreamer2 @ e998419b

-Subproject commit 6250875949495a354aea7f62e1f7aa9652397f84
+Subproject commit e998419b7fd1e43a39974d2cf408ae169888a62b

tester/CMakeLists.txt

@@ -22,6 +22,7 @@
 
 set(SOURCE_FILES
 	accountmanager.c
+	audio_bypass_tester.c
 	call_tester.c
 	complex_sip_call_tester.c
 	dtmf_tester.c

tester/Makefile.am

@@ -112,6 +112,7 @@ lib_LTLIBRARIES = liblinphonetester.la
 
 liblinphonetester_la_SOURCES = \
 				accountmanager.c \
+				audio_bypass_tester.c \
 				call_tester.c \
 				complex_sip_call_tester.c \
 				dtmf_tester.c \

tester/audio_bypass_tester.c

+/*
+	liblinphone_tester - liblinphone test suite
+	Copyright (C) 2013  Belledonne Communications SARL
+
+	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, see <http://www.gnu.org/licenses/>.
+*/
+
+#include "liblinphone_tester.h"
+#include "private.h"
+#include "mediastreamer2/msfileplayer.h"
+#include "mediastreamer2/msfilerec.h"
+
+static void audio_bypass_snd_read_init(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_read_preprocess(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_read_process(MSFilter *f) {
+	mblk_t *m = allocb(10, 0);
+	memset(m->b_wptr, 0, 10);
+	m->b_wptr += 10;
+	ms_queue_put(f->outputs[0], m);
+}
+
+static void audio_bypass_snd_read_postprocess(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_read_uninit(MSFilter *f) {
+	
+}
+
+static int audio_bypass_snd_read_get_sample_rate(MSFilter *f, void *arg) {
+	int *sample_rate = (int *)arg;
+	*sample_rate = 44100;
+	return 0;
+}
+
+static int audio_bypass_snd_read_get_nchannels(MSFilter *f, void *arg) {
+	int *nchannels = (int *)arg;
+	*nchannels = 1;
+	return 0;
+}
+
+static int audio_bypass_snd_read_get_fmt(MSFilter *f, void *arg) {
+	MSPinFormat *pinFmt = (MSPinFormat *)arg;
+	pinFmt->fmt = ms_factory_get_audio_format(f->factory, "L16", 44100, 1, NULL);
+	return 0;
+}
+
+static MSFilterMethod audio_bypass_snd_read_methods[] = {
+	{ MS_FILTER_GET_SAMPLE_RATE, audio_bypass_snd_read_get_sample_rate },
+	{ MS_FILTER_GET_NCHANNELS, audio_bypass_snd_read_get_nchannels },
+	{ MS_FILTER_GET_OUTPUT_FMT, audio_bypass_snd_read_get_fmt },
+	{ 0, NULL }
+};
+
+MSFilterDesc audio_bypass_snd_read_desc = {
+	MS_FILTER_PLUGIN_ID,
+	"AudioBypassReader",
+	"audio bypass source",
+	MS_FILTER_OTHER,
+	NULL,
+	0,
+	1,
+	audio_bypass_snd_read_init,
+	audio_bypass_snd_read_preprocess,
+	audio_bypass_snd_read_process,
+	audio_bypass_snd_read_postprocess,
+	audio_bypass_snd_read_uninit,
+	audio_bypass_snd_read_methods
+};
+
+static void audio_bypass_snd_write_init(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_write_preprocess(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_write_process(MSFilter *f) {
+	ms_queue_get(f->inputs[0]);	
+}
+
+static void audio_bypass_snd_write_postprocess(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_write_uninit(MSFilter *f) {
+	
+}
+
+static int audio_bypass_snd_write_get_sample_rate(MSFilter *f, void *arg) {
+	int *sample_rate = (int*)arg;
+	*sample_rate = 44100;
+	return 0;
+}
+
+static int audio_bypass_snd_write_get_nchannels(MSFilter *obj, void *arg) {
+	int *nchannels = (int*)arg;
+	*nchannels = 1;
+	return 0;
+}
+
+static int audio_bypass_snd_write_get_fmt(MSFilter *f, void *arg) {
+	MSPinFormat *pinFmt = (MSPinFormat *)arg;
+	pinFmt->fmt = ms_factory_get_audio_format(f->factory, "L16", 44100, 1, NULL);
+	return 0;
+}
+
+static MSFilterMethod audio_bypass_snd_write_methods[] = {
+	{ MS_FILTER_GET_SAMPLE_RATE, audio_bypass_snd_write_get_sample_rate },
+	{ MS_FILTER_GET_NCHANNELS, audio_bypass_snd_write_get_nchannels },
+	{ MS_FILTER_GET_OUTPUT_FMT,	audio_bypass_snd_write_get_fmt },
+	{ 0, NULL }
+};
+
+MSFilterDesc audio_bypass_snd_write_desc = {
+	MS_FILTER_PLUGIN_ID,
+	"AudioBypassWriter",
+	"audio bypass output",
+	MS_FILTER_OTHER,
+	NULL,
+	1,
+	0,
+	audio_bypass_snd_write_init,
+	audio_bypass_snd_write_preprocess,
+	audio_bypass_snd_write_process,
+	audio_bypass_snd_write_postprocess,
+	audio_bypass_snd_write_uninit,
+	audio_bypass_snd_write_methods
+};
+
+static MSFilter* audio_bypass_snd_card_create_reader(MSSndCard *sndcard) {
+	MSFactory *factory = ms_snd_card_get_factory(sndcard);
+	MSFilter *f = ms_factory_create_filter_from_desc(factory, &audio_bypass_snd_read_desc);
+	return f;
+}
+
+static MSFilter* audio_bypass_snd_card_create_writer(MSSndCard *sndcard) {
+	MSFactory *factory = ms_snd_card_get_factory(sndcard);
+	MSFilter *f = ms_factory_create_filter_from_desc(factory, &audio_bypass_snd_write_desc);
+	
+	MSPinFormat *pinfmt = ms_new0(MSPinFormat, 0);
+	pinfmt->pin = 0;
+	pinfmt->fmt = ms_factory_get_audio_format(factory, "L16", 44100, 1, NULL);
+	
+	ms_filter_call_method(f, MS_FILTER_SET_OUTPUT_FMT, pinfmt);
+	ms_free(pinfmt);
+	
+	return f;
+}
+
+static void audio_bypass_snd_card_detect(MSSndCardManager *m);
+
+MSSndCardDesc audio_bypass_snd_card_desc = {
+	"audioBypass",
+	audio_bypass_snd_card_detect,
+	NULL,
+	NULL,
+	NULL,
+	NULL,
+	NULL,
+	NULL,	
+	audio_bypass_snd_card_create_reader,
+	audio_bypass_snd_card_create_writer,
+	NULL
+};
+
+static MSSndCard* create_audio_bypass_snd_card(void) {
+	MSSndCard* sndcard;
+	sndcard = ms_snd_card_new(&audio_bypass_snd_card_desc);
+	sndcard->data = NULL;
+	sndcard->name = ms_strdup("audio bypass sound card");
+	sndcard->capabilities = MS_SND_CARD_CAP_PLAYBACK | MS_SND_CARD_CAP_CAPTURE;
+	sndcard->latency = 0;
+	return sndcard;
+}
+
+static void audio_bypass_snd_card_detect(MSSndCardManager *m) {
+	ms_snd_card_manager_add_card(m, create_audio_bypass_snd_card());
+}
+
+static void audio_bypass(void) {
+	LinphoneCoreManager *marie = linphone_core_manager_new("marie_rc_audio_bypass");
+	LinphoneCore *marie_lc = marie->lc;
+	MSFactory *marie_factory = linphone_core_get_ms_factory(marie_lc);
+	MSSndCardManager *marie_sndcard_manager = ms_factory_get_snd_card_manager(marie_factory);
+	
+	LinphoneCoreManager *pauline = linphone_core_manager_new("pauline_rc_audio_bypass");
+	LinphoneCore *pauline_lc = pauline->lc;
+	MSFactory *pauline_factory = linphone_core_get_ms_factory(pauline_lc);
+	MSSndCardManager *pauline_sndcard_manager = ms_factory_get_snd_card_manager(pauline_factory);
+	
+	bool_t call_ok;
+	MSList *marie_audio_codecs = marie_lc->codecs_conf.audio_codecs;
+	MSList *pauline_audio_codecs = pauline_lc->codecs_conf.audio_codecs;
+	
+	// Enable L16 audio codec
+	while (marie_audio_codecs) {
+		PayloadType *pt = (PayloadType *)marie_audio_codecs->data;
+		if (strcmp(pt->mime_type, "L16") == 0 && pt->channels == 1) {
+			pt->flags |= PAYLOAD_TYPE_ENABLED;
+		} else {
+			pt->flags &= PAYLOAD_TYPE_ENABLED;
+		}
+		marie_audio_codecs = ms_list_next(marie_audio_codecs);
+	}
+	while (pauline_audio_codecs) {
+		PayloadType *pt = (PayloadType *)pauline_audio_codecs->data;
+		if (strcmp(pt->mime_type, "L16") == 0 && pt->channels == 1) {
+			pt->flags |= PAYLOAD_TYPE_ENABLED;
+		} else {
+			pt->flags &= PAYLOAD_TYPE_ENABLED;
+		}
+		pauline_audio_codecs = ms_list_next(pauline_audio_codecs);
+	}
+	
+	// Add our custom sound card
+	ms_snd_card_manager_register_desc(marie_sndcard_manager, &audio_bypass_snd_card_desc);
+	ms_snd_card_manager_register_desc(pauline_sndcard_manager, &audio_bypass_snd_card_desc);
+	linphone_core_reload_sound_devices(marie->lc);
+	linphone_core_reload_sound_devices(pauline->lc);
+	linphone_core_set_playback_device(marie->lc, "audioBypass: audio bypass sound card");
+	linphone_core_set_playback_device(pauline->lc, "audioBypass: audio bypass sound card");
+	linphone_core_set_capture_device(marie->lc, "audioBypass: audio bypass sound card");
+	linphone_core_set_capture_device(pauline->lc, "audioBypass: audio bypass sound card");
+
+	call_ok = call(marie, pauline);
+	BC_ASSERT_TRUE(call_ok);
+	if (!call_ok) goto end;
+	
+	BC_ASSERT_STRING_EQUAL(linphone_call_params_get_used_audio_codec(linphone_call_get_current_params(linphone_core_get_current_call(marie_lc)))->mime_type, "L16");
+	
+	wait_for_until(pauline->lc, marie->lc, NULL, 0, 10000);
+	end_call(marie, pauline);
+	
+end:
+	linphone_core_manager_destroy(marie);
+	linphone_core_manager_destroy(pauline);
+}
+
+test_t audio_bypass_tests[] = {
+	TEST_NO_TAG("Audio Bypass", audio_bypass)
+};
+
+test_suite_t audio_bypass_suite = { "Audio Bypass", NULL, NULL, 
+	liblinphone_tester_before_each, liblinphone_tester_after_each,
+	sizeof(audio_bypass_tests) / sizeof(audio_bypass_tests[0]), audio_bypass_tests };

tester/liblinphone_tester.h

@@ -61,6 +61,7 @@ extern test_suite_t multicast_call_test_suite;
 extern test_suite_t multi_call_test_suite;
 extern test_suite_t proxy_config_test_suite;
 extern test_suite_t vcard_test_suite;
+extern test_suite_t audio_bypass_suite;
 #if HAVE_SIPP
 extern test_suite_t complex_sip_call_test_suite;
 #endif

tester/rcfiles/marie_rc_audio_bypass

+[sip]
+sip_port=-1
+sip_tcp_port=-1
+sip_tls_port=-1
+default_proxy=0
+ping_with_options=0
+composing_idle_timeout=1
+store_ha1_passwd=0 #used for sipp
+
+[auth_info_0]
+username=marie
+userid=marie
+passwd=secret
+realm=sip.example.org
+
+[proxy_0]
+reg_proxy=sip.example.org;transport=tcp
+reg_route=sip.example.org;transport=tcp;lr
+reg_identity="Super Marie" <sip:marie@sip.example.org>
+reg_expires=3600
+reg_sendregister=1
+publish=0
+dial_escape_plus=0
+
+[friend_0]
+url="Paupoche" <sip:pauline@sip.example.org>
+pol=accept
+subscribe=0
+
+[rtp]
+audio_rtp_port=18070-28000
+video_rtp_port=28070-38000
+text_rtp_port=39000-49000
+
+[video]
+display=0
+capture=0
+show_local=0
+size=qcif
+enabled=0
+self_view=0
+automatically_initiate=0
+automatically_accept=0
+device=StaticImage: Static picture
+
+[sound]
+echocancellation=0 #to not overload cpu in case of VG
+features=NONE
+
+[net]
+dns_srv_enabled=0 #no srv needed in general
+stun_server=stun.linphone.org
+
+[misc]
+add_missing_audio_codecs=0
+
+[audio_codec_0]
+mime=L16
+rate=44100
+channels=2
+enabled=0
+
+[audio_codec_1]
+mime=L16
+rate=44100
+channels=1
+enabled=1

tester/rcfiles/pauline_rc_audio_bypass

+[sip]
+sip_port=-1
+sip_tcp_port=-1
+sip_tls_port=-1
+default_proxy=0
+ping_with_options=0
+composing_idle_timeout=1
+
+[auth_info_0]
+username=pauline
+userid=pauline
+passwd=secret
+realm=sip.example.org
+
+[proxy_0]
+reg_proxy=sip2.linphone.org;transport=tls
+reg_route=sip2.linphone.org;transport=tls
+reg_identity=sip:pauline@sip.example.org
+reg_expires=3600
+reg_sendregister=1
+publish=0
+dial_escape_plus=0
+
+#[friend_0]
+#url="Mariette" <sip:marie@sip.example.org>
+#pol=accept
+#subscribe=0
+
+[rtp]
+audio_rtp_port=18070-28000
+video_rtp_port=39072-49000
+
+[video]
+display=0
+capture=0
+show_local=0
+size=qcif
+enabled=0
+self_view=0
+automatically_initiate=0
+automatically_accept=0
+device=StaticImage: Static picture
+
+[sound]
+echocancellation=0 #to not overload cpu in case of VG
+features=NONE
+
+[net]
+dns_srv_enabled=0 #no srv needed in general
+stun_server=stun.linphone.org
+
+[misc]
+add_missing_audio_codecs=0
+
+[audio_codec_0]
+mime=L16
+rate=44100
+channels=2
+enabled=0
+
+[audio_codec_1]
+mime=L16
+rate=44100
+channels=1
+enabled=1

tester/tester.c

@@ -480,6 +480,7 @@ void liblinphone_tester_add_suites() {
 	bc_tester_add_suite(&tunnel_test_suite);
 	bc_tester_add_suite(&offeranswer_test_suite);
 	bc_tester_add_suite(&call_test_suite);
+	bc_tester_add_suite(&audio_bypass_suite);
 	bc_tester_add_suite(&multi_call_test_suite);
 	bc_tester_add_suite(&message_test_suite);
 	bc_tester_add_suite(&presence_test_suite);