Merge branch 'dev_audio_bypass'

mediastreamer2

@@ -1 +1 @@
-Subproject commit c01ab3c2df96a2aad53edde9b4759344ed3a5bad
+Subproject commit 128815537044ea17808a0a5843b2762364f8203d

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

@@ -1,6 +1,7 @@
 
 TESTER_SOUNDS = sounds/ahbahouaismaisbon.wav \
 				sounds/hello8000.wav  \
+				sounds/hello44100.wav  \
 				sounds/oldphone.wav \
 				sounds/sintel_trailer_opus_h264.mkv \
 				sounds/sintel_trailer_pcmu_h264.mkv \
@@ -112,6 +113,7 @@ 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

@@ -0,0 +1,501 @@
+/*
+	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 "audio_bypass_wav_header.h" // This is a copy of mediastreamer2/src/audiofilters/wav_header.h
+
+/**********************************************************************
+ * This is a (simpler) copy of msfileplay filter in mediastreamer2   *
+ *********************************************************************/
+
+struct _PlayerData{
+	int fd;
+	MSPlayerState state;
+	int rate;
+	int nchannels;
+	int hsize;
+	int loop_after;
+	int pause_time;
+	int count;
+	int samplesize;
+	char *mime;
+	uint32_t ts;
+	bool_t swap;
+	bool_t is_raw;
+};
+
+typedef struct _PlayerData PlayerData;
+
+static void audio_bypass_snd_read_init(MSFilter *f) {
+	PlayerData *d=ms_new0(PlayerData,1);
+	d->fd=-1;
+	d->state=MSPlayerClosed;
+	d->swap=TRUE;
+	d->rate=44100;
+	d->nchannels=1;
+	d->samplesize=2;
+	d->mime = "L16";
+	d->hsize=0;
+	d->loop_after=-1; /*by default, don't loop*/
+	d->pause_time=0;
+	d->count=0;
+	d->ts=0;
+	d->is_raw=TRUE;
+	f->data=d;
+}
+
+int ms_read_wav_header_from_fd(wave_header_t *header,int fd){
+	int count;
+	int skip;
+	int hsize=0;
+	riff_t *riff_chunk=&header->riff_chunk;
+	format_t *format_chunk=&header->format_chunk;
+	data_t *data_chunk=&header->data_chunk;
+	
+	unsigned long len=0;
+	
+	len = read(fd, (char*)riff_chunk, sizeof(riff_t)) ;
+	if (len != sizeof(riff_t)){
+		goto not_a_wav;
+	}
+	
+	if (0!=strncmp(riff_chunk->riff, "RIFF", 4) || 0!=strncmp(riff_chunk->wave, "WAVE", 4)){
+		goto not_a_wav;
+	}
+	
+	len = read(fd, (char*)format_chunk, sizeof(format_t)) ;            
+	if (len != sizeof(format_t)){
+		ms_warning("Wrong wav header: cannot read file");
+		goto not_a_wav;
+	}
+	
+	if ((skip=le_uint32(format_chunk->len)-0x10)>0)
+	{
+		lseek(fd,skip,SEEK_CUR);
+	}
+	hsize=sizeof(wave_header_t)-0x10+le_uint32(format_chunk->len);
+	
+	count=0;
+	do{
+		len = read(fd, data_chunk, sizeof(data_t)) ;
+		if (len != sizeof(data_t)){
+			ms_warning("Wrong wav header: cannot read file");
+			goto not_a_wav;
+		}
+		if (strncmp(data_chunk->data, "data", 4)!=0){
+			ms_warning("skipping chunk=%c%c%c%c len=%i", data_chunk->data[0],data_chunk->data[1],data_chunk->data[2],data_chunk->data[3], data_chunk->len);
+			lseek(fd,le_uint32(data_chunk->len),SEEK_CUR);
+			count++;
+			hsize+=len+le_uint32(data_chunk->len);
+		}else{
+			hsize+=len;
+			break;
+		}
+	}while(count<30);
+	return hsize;
+
+	not_a_wav:
+		/*rewind*/
+		lseek(fd,0,SEEK_SET);
+		return -1;
+}
+
+static int read_wav_header(PlayerData *d){
+	wave_header_t header;
+	format_t *format_chunk=&header.format_chunk;
+	int ret=ms_read_wav_header_from_fd(&header,d->fd);
+	
+	d->samplesize=le_uint16(format_chunk->blockalign)/d->nchannels;
+	d->hsize=ret;
+	
+	#ifdef WORDS_BIGENDIAN
+	if (le_uint16(format_chunk->blockalign)==le_uint16(format_chunk->channel) * 2)
+		d->swap=TRUE;
+	#endif
+	d->is_raw=FALSE;
+	return 0;
+}
+
+static void audio_bypass_snd_read_preprocess(MSFilter *f) {
+	PlayerData *d=(PlayerData*)f->data;
+	int fd;
+	const char *file=bc_tester_res("sounds/hello44100.wav");
+
+	if ((fd=open(file,O_RDONLY|O_BINARY))==-1){
+		ms_warning("MSFilePlayer[%p]: failed to open %s: %s",f,file,strerror(errno));
+		return;
+	}
+
+	d->state=MSPlayerPaused;
+	d->fd=fd;
+	d->ts=0;
+	if (read_wav_header(d)!=0 && strstr(file,".wav")){
+		ms_warning("File %s has .wav extension but wav header could be found.",file);
+	}
+	ms_filter_notify_no_arg(f,MS_FILTER_OUTPUT_FMT_CHANGED);
+	ms_message("MSFilePlayer[%p]: %s opened: rate=%i,channel=%i",f,file,d->rate,d->nchannels);
+	
+	if (d->state==MSPlayerPaused)
+		d->state=MSPlayerPlaying;
+	return;
+}
+
+static void swap_bytes(unsigned char *bytes, int len){
+	int i;
+	unsigned char tmp;
+	for(i=0;i<len;i+=2){
+		tmp=bytes[i];
+		bytes[i]=bytes[i+1];
+		bytes[i+1]=tmp;
+	}
+}
+
+static void audio_bypass_snd_read_process(MSFilter *f) {
+	PlayerData *d=(PlayerData*)f->data;
+	int nsamples=(f->ticker->interval*d->rate*d->nchannels)/1000;
+	int bytes;
+	/*send an even number of samples each tick. At 22050Hz the number of samples per 10 ms chunk is odd.
+	Odd size buffer of samples cause troubles to alsa. Fixing in alsa is difficult, so workaround here.
+	*/
+	if (nsamples & 0x1 ) { //odd number of samples
+		if (d->count & 0x1 )
+			nsamples++;
+		else
+			nsamples--;
+	}
+	bytes=nsamples*d->samplesize;
+	d->count++;
+	ms_filter_lock(f);
+	if (d->state==MSPlayerPlaying){
+		{
+			int err;
+			mblk_t *om=allocb(bytes,0);
+			if (d->pause_time>0){
+				err=bytes;
+				memset(om->b_wptr,0,bytes);
+				d->pause_time-=f->ticker->interval;
+			}else{
+				err=read(d->fd,om->b_wptr,bytes);
+				if (d->swap) swap_bytes(om->b_wptr,bytes);
+			}
+			if (err>=0){
+				if (err!=0){
+					if (err<bytes)
+						memset(om->b_wptr+err,0,bytes-err);
+					om->b_wptr+=bytes;
+					mblk_set_timestamp_info(om,d->ts);
+					d->ts+=nsamples;
+					ms_queue_put(f->outputs[0],om);
+				}else freemsg(om);
+				if (err<bytes){
+					ms_filter_notify_no_arg(f,MS_PLAYER_EOF);
+					/*for compatibility:*/
+					lseek(d->fd,d->hsize,SEEK_SET);
+
+					/* special value for playing file only once */
+					if (d->loop_after<0)
+					{
+						d->state=MSPlayerPaused;
+						ms_filter_unlock(f);
+						return;
+					}
+
+					if (d->loop_after>=0){
+						d->pause_time=d->loop_after;
+					}
+				}
+			}else{
+				ms_warning("Fail to read %i bytes: %s",bytes,strerror(errno));
+			}
+		}
+	}
+	ms_filter_unlock(f);
+}
+
+static void audio_bypass_snd_read_postprocess(MSFilter *f) {
+	PlayerData *d=(PlayerData*)f->data;
+	ms_filter_lock(f);
+	if (d->state!=MSPlayerClosed){
+		d->state=MSPlayerPaused;
+		lseek(d->fd,d->hsize,SEEK_SET);
+	}
+	ms_filter_unlock(f);
+	if (d->fd!=-1)	close(d->fd);
+	d->fd=-1;
+	d->state=MSPlayerClosed;
+}
+
+static void audio_bypass_snd_read_uninit(MSFilter *f) {
+	PlayerData *d=(PlayerData*)f->data;
+	ms_free(d);
+}
+
+static int audio_bypass_snd_read_set_sample_rate(MSFilter *f, void *arg) { // This is to prevent ms2 to put a resampler between this filter and the rtpsend
+	return 0;
+}
+
+static int audio_bypass_snd_read_set_nchannels(MSFilter *f, void *arg) { // This is to prevent ms2 to put a resampler between this filter and the rtpsend
+	return 0;
+}
+
+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_SET_SAMPLE_RATE, audio_bypass_snd_read_set_sample_rate },
+	{ MS_FILTER_SET_NCHANNELS, audio_bypass_snd_read_set_nchannels },
+	{ 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) {
+	mblk_t *m = ms_queue_get(f->inputs[0]);
+	ms_free(m);
+}
+
+static void audio_bypass_snd_write_postprocess(MSFilter *f) {
+	
+}
+
+static void audio_bypass_snd_write_uninit(MSFilter *f) {
+	
+}
+
+static int audio_bypass_snd_write_set_sample_rate(MSFilter *f, void *arg) { // This is to prevent ms2 to put a resampler between this filter and the rtprecv
+	return 0;
+}
+
+static int audio_bypass_snd_write_set_nchannels(MSFilter *f, void *arg) { // This is to prevent ms2 to put a resampler between this filter and the rtprecv
+	return 0;
+}
+
+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_SET_SAMPLE_RATE, audio_bypass_snd_write_set_sample_rate },
+	{ MS_FILTER_SET_NCHANNELS, audio_bypass_snd_write_set_nchannels },
+	{ 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);
+	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;
+}
+
+#define AUDIO_BYPASS_SOUNDCARD "audioBypass: audio bypass sound card"
+
+static void audio_bypass_snd_card_detect(MSSndCardManager *m) {
+	ms_snd_card_manager_add_card(m, create_audio_bypass_snd_card());
+}
+
+static void only_enable_payload(LinphoneCore *lc, const char *mime, int rate, int channels) {
+	const MSList *elem = linphone_core_get_audio_codecs(lc);
+	PayloadType *pt;
+
+	for(; elem != NULL; elem = elem->next) {
+		pt = (PayloadType *)elem->data;
+		linphone_core_enable_payload_type(lc, pt, FALSE);
+	}
+	pt = linphone_core_find_payload_type(lc, mime, rate, channels);
+	if (BC_ASSERT_PTR_NOT_NULL(pt)) {
+		linphone_core_enable_payload_type(lc, pt, TRUE);
+	}
+}
+
+static void audio_bypass(void) {
+	LinphoneCoreManager *marie = linphone_core_manager_new("marie_rc");
+	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");
+	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;
+	char *hellopath = bc_tester_res("sounds/hello44100.wav");
+	char *recordpath = bc_tester_file("audiobypass-record.wav");
+	double similar=1;
+	const double threshold = 0.85;
+	
+	lp_config_set_string(marie_lc->config, "sound", "features", "None");
+	lp_config_set_string(pauline_lc->config, "sound", "features", "None");
+	
+	/*make sure the record file doesn't already exists, otherwise this test will append new samples to it*/
+	unlink(recordpath);
+	
+	// Enable L16 audio codec
+	only_enable_payload(marie_lc, "L16", 44100, 1);
+	only_enable_payload(pauline_lc, "L16", 44100, 1);
+	
+	// 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, AUDIO_BYPASS_SOUNDCARD);
+	linphone_core_set_playback_device(pauline_lc, AUDIO_BYPASS_SOUNDCARD);
+	linphone_core_set_capture_device(marie_lc, AUDIO_BYPASS_SOUNDCARD);
+	linphone_core_set_capture_device(pauline_lc, AUDIO_BYPASS_SOUNDCARD);
+	BC_ASSERT_STRING_EQUAL(linphone_core_get_capture_device(marie_lc), AUDIO_BYPASS_SOUNDCARD);
+	BC_ASSERT_STRING_EQUAL(linphone_core_get_capture_device(pauline_lc), AUDIO_BYPASS_SOUNDCARD);
+	BC_ASSERT_STRING_EQUAL(linphone_core_get_playback_device(marie_lc), AUDIO_BYPASS_SOUNDCARD);
+	BC_ASSERT_STRING_EQUAL(linphone_core_get_playback_device(pauline_lc), AUDIO_BYPASS_SOUNDCARD);
+
+	linphone_core_use_files(pauline_lc, TRUE);
+	linphone_core_set_play_file(pauline_lc, NULL);
+	linphone_core_set_record_file(pauline_lc, recordpath);
+
+	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, 22000); //hello44100.wav is 22 seconds long
+	end_call(marie, pauline);
+	
+	BC_ASSERT_EQUAL(ms_audio_diff(hellopath, recordpath, &similar, &audio_cmp_params, NULL, NULL), 0, int, "%d");
+	BC_ASSERT_GREATER(similar, threshold, double, "%g");
+	BC_ASSERT_LOWER(similar, 1.0, double, "%g");
+
+end:
+	bc_free(recordpath);
+	bc_free(hellopath);
+	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/audio_bypass_wav_header.h

@@ -0,0 +1,139 @@
+/*
+mediastreamer2 library - modular sound and video processing and streaming
+Copyright (C) 2006  Simon MORLAT (simon.morlat@linphone.org)
+
+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.
+*/
+
+
+#ifndef waveheader_h
+#define waveheader_h
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
+#ifdef _WIN32
+#	include <io.h>
+#	ifndef R_OK
+#		define R_OK 0x2
+#	endif
+#	ifndef W_OK
+#		define W_OK 0x6
+#	endif
+#   ifndef F_OK
+#       define F_OK 0x0
+#   endif
+
+#	ifndef S_IRUSR
+#	define S_IRUSR S_IREAD
+#	endif
+
+#	ifndef S_IWUSR
+#	define S_IWUSR S_IWRITE
+#	endif
+
+#	define open _open
+#	define read _read
+#	define write _write
+#	define close _close
+#	define access _access
+#	define lseek _lseek
+#else /*_WIN32*/
+
+#	ifndef O_BINARY
+#	define O_BINARY 0
+#	endif
+
+#endif /*!_WIN32*/
+
+#ifdef swap16
+#else
+/* all integer in wav header must be read in least endian order */
+static MS2_INLINE uint16_t swap16(uint16_t a)
+{
+	return ((a & 0xFF) << 8) | ((a & 0xFF00) >> 8);
+}
+#endif
+
+#ifdef swap32
+#else
+static MS2_INLINE uint32_t swap32(uint32_t a)
+{
+	return ((a & 0xFF) << 24) | ((a & 0xFF00) << 8) |
+		((a & 0xFF0000) >> 8) | ((a & 0xFF000000) >> 24);
+}
+#endif
+
+#ifdef WORDS_BIGENDIAN
+#define le_uint32(a) (swap32((a)))
+#define le_uint16(a) (swap16((a)))
+#define le_int16(a) ( (int16_t) swap16((uint16_t)((a))) )
+#else
+#define le_uint32(a) (a)
+#define le_uint16(a) (a)
+#define le_int16(a) (a)
+#endif
+
+typedef struct _riff_t {
+	char riff[4] ;	/* "RIFF" (ASCII characters) */
+	uint32_t  len ;	/* Length of package (binary, little endian) */
+	char wave[4] ;	/* "WAVE" (ASCII characters) */
+} riff_t;
+
+/* The FORMAT chunk */
+
+typedef struct _format_t {
+	char  fmt[4] ;		/* "fmt_" (ASCII characters) */
+	uint32_t   len ;	/* length of FORMAT chunk (always 0x10) */
+	uint16_t  type;		/* codec type*/
+	uint16_t channel ;	/* Channel numbers (0x01 = mono, 0x02 = stereo) */
+	uint32_t   rate ;	/* Sample rate (binary, in Hz) */
+	uint32_t   bps ;	/* Average Bytes Per Second */
+	uint16_t blockalign ;	/*number of bytes per sample */
+	uint16_t bitpspl ;	/* bits per sample */
+} format_t;
+
+/* The DATA chunk */
+
+typedef struct _data_t {
+	char data[4] ;	/* "data" (ASCII characters) */
+	uint32_t  len ;	/* length of data */
+} data_t;
+
+typedef struct _wave_header_t
+{
+	riff_t riff_chunk;
+	format_t format_chunk;
+	data_t data_chunk;
+} wave_header_t;
+
+#ifndef _WIN32
+#define WAVE_FORMAT_PCM			0x0001
+#define WAVE_FORMAT_IEEE_FLOAT	0x0003
+#define WAVE_FORMAT_ALAW		0x0006
+#define WAVE_FORMAT_MULAW		0x0007
+#define WAVE_FORMAT_EXTENSIBLE	0xFFFE
+#endif
+
+#define wave_header_get_format_type(header)	le_uint16((header)->format_chunk.type)
+#define wave_header_get_rate(header)		le_uint32((header)->format_chunk.rate)
+#define wave_header_get_channel(header)		le_uint16((header)->format_chunk.channel)
+#define wave_header_get_bpsmpl(header) \
+	le_uint16((header)->format_chunk.blockalign)
+
+int ms_read_wav_header_from_fd(wave_header_t *header,int fd);
+
+#endif

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/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);