static void adaptive_opus_audio_stream() {
bool_t supported = ms_filter_codec_supported("opus");
if( supported ) {
// at 8KHz -> 24kb/s
// at 48KHz -> 48kb/s
float bw_usage;
stream_manager_t * marielle, * margaux;
// on EDGEBW, both should be overconsumming
start_adaptive_stream(AudioStreamType, &marielle, &margaux, OPUS_PAYLOAD_TYPE, 8000, EDGE_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./EDGE_BW;
CU_ASSERT_IN_RANGE(bw_usage, 2.f, 3.f); // bad! since this codec cant change its ptime and it is the lower bitrate, no improvement can occur
stop_adaptive_stream(marielle,margaux);
start_adaptive_stream(AudioStreamType, &marielle, &margaux, OPUS_PAYLOAD_TYPE, 48000, EDGE_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./EDGE_BW;
CU_ASSERT_IN_RANGE(bw_usage, 1.f, 1.4f); // bad!
stop_adaptive_stream(marielle,margaux);
// on 3G BW, both should be at max
start_adaptive_stream(AudioStreamType, &marielle, &margaux, OPUS_PAYLOAD_TYPE, 8000, THIRDGENERATION_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./THIRDGENERATION_BW;
CU_ASSERT_IN_RANGE(bw_usage, .1f, .15f);
stop_adaptive_stream(marielle,margaux);
start_adaptive_stream(AudioStreamType, &marielle, &margaux, OPUS_PAYLOAD_TYPE, 48000, THIRDGENERATION_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./THIRDGENERATION_BW;
CU_ASSERT_IN_RANGE(bw_usage, .2f, .3f);
stop_adaptive_stream(marielle,margaux);
}
}
static void iterate_adaptive_stream(stream_manager_t * marielle, stream_manager_t * margaux,
int timeout_ms, int* current, int expected){
int retry=0;
MediaStream *marielle_ms,*margaux_ms;
if (marielle->type == AudioStreamType){
marielle_ms=&marielle->audio_stream->ms;
margaux_ms=&margaux->audio_stream->ms;
}else{
marielle_ms=&marielle->video_stream->ms;
margaux_ms=&margaux->video_stream->ms;
}
while ((!current||*current<expected) && retry++ <timeout_ms/100) {
media_stream_iterate(marielle_ms);
media_stream_iterate(margaux_ms);
handle_queue_events(marielle);
if (retry%10==0) {
ms_message("stream [%p] bandwidth usage: [d=%.1f,u=%.1f] kbit/sec" ,
marielle_ms, media_stream_get_down_bw(marielle_ms)/1000, media_stream_get_up_bw(marielle_ms)/1000);
ms_message("stream [%p] bandwidth usage: [d=%.1f,u=%.1f] kbit/sec" ,
margaux_ms, media_stream_get_down_bw(margaux_ms)/1000, media_stream_get_up_bw(margaux_ms)/1000);
}
ms_usleep(100000);
}
}
bool_t wait_for_list_with_parse_events(MSList *mss, int *counter, int value, int timeout_ms, MSList *cbs, MSList *ptrs) {
MSList *msi;
MSList *cbi;
MSList *ptri;
int retry = 0;
while ((*counter < value) && (retry++ < (timeout_ms / 100))) {
for (msi = mss, cbi = cbs, ptri = ptrs; msi != NULL; msi = msi->next) {
MediaStream *stream = (MediaStream *)msi->data;
ms_tester_iterate_cb cb = NULL;
media_stream_iterate(stream);
if ((retry % 10) == 0) {
ms_message("stream [%p] bandwidth usage: [d=%.1f,u=%.1f] kbit/sec",
stream,
media_stream_get_down_bw(stream) / 1000,
media_stream_get_up_bw(stream) / 1000);
}
if (cbi && ptri) {
cb = (ms_tester_iterate_cb)cbi->data;
cb(stream, ptri->data);
}
if (cbi) cbi = cbi->next;
if (ptri) ptri = ptri->next;
}
ms_usleep(100000);
}
if (*counter < value)
return FALSE;
return TRUE;
}
static float adaptive_audio_stream(int codec_payload, int initial_bitrate,int target_bw, int max_recv_rtcp_packet) {
stream_manager_t * marielle = stream_manager_new();
stream_manager_t * margaux = stream_manager_new();
int pause_time=0;
OrtpNetworkSimulatorParams params={0};
params.enabled=TRUE;
params.loss_rate=0;
params.max_bandwidth=target_bw;
params.max_buffer_size=initial_bitrate;
float bw_usage_ratio;
float marielle_send_bw;
media_stream_enable_adaptive_bitrate_control(&marielle->stream->ms,TRUE);
stream_manager_start(marielle,codec_payload, margaux->local_rtp,initial_bitrate,HELLO_16K_1S_FILE,NULL);
ms_filter_call_method(marielle->stream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
stream_manager_start(margaux,codec_payload, marielle->local_rtp,-1,NULL,RECORDED_16K_1S_FILE);
rtp_session_enable_network_simulation(margaux->stream->ms.sessions.rtp_session,¶ms);
wait_for_until(&marielle->stream->ms,&margaux->stream->ms,&marielle->stats.number_of_EndOfFile,10,2500*max_recv_rtcp_packet);
marielle_send_bw=media_stream_get_up_bw(&marielle->stream->ms);
bw_usage_ratio=marielle_send_bw/params.max_bandwidth;
ms_message("marielle sent bw=[%f], target was [%f] bw_usage_ratio [%f]",marielle_send_bw,params.max_bandwidth,bw_usage_ratio);
stream_manager_delete(marielle);
stream_manager_delete(margaux);
unlink(RECORDED_16K_1S_FILE);
return bw_usage_ratio;
}
static void adaptive_pcma_audio_stream() {
bool_t supported = ms_filter_codec_supported("pcma");
if( supported ) {
// at 8KHz -> 80 kb/s
float bw_usage;
stream_manager_t * marielle, * margaux;
// yet non-adaptative codecs cannot respect low throughput limitations
start_adaptive_stream(AudioStreamType, &marielle, &margaux, PCMA8_PAYLOAD_TYPE, 8000, EDGE_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./EDGE_BW;
CU_ASSERT_IN_RANGE(bw_usage,6.f, 8.f); // this is bad!
stop_adaptive_stream(marielle,margaux);
start_adaptive_stream(AudioStreamType, &marielle, &margaux, PCMA8_PAYLOAD_TYPE, 8000, THIRDGENERATION_BW, 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./THIRDGENERATION_BW;
CU_ASSERT_IN_RANGE(bw_usage, .3f, .5f);
stop_adaptive_stream(marielle,margaux);
}
}
static void adaptive_speex16_audio_stream() {
bool_t supported = ms_filter_codec_supported("speex");
if( supported ) {
// at 16KHz -> 20 kb/s
// at 32KHz -> 30 kb/s
float bw_usage;
stream_manager_t * marielle, * margaux;
start_adaptive_stream(AudioStreamType, &marielle, &margaux, SPEEX16_PAYLOAD_TYPE, 32000, EDGE_BW / 2., 0, 0, 0);
iterate_adaptive_stream(marielle, margaux, 10000, NULL, 0);
bw_usage=media_stream_get_up_bw(&marielle->audio_stream->ms)*1./(EDGE_BW / 2.);
CU_ASSERT_IN_RANGE(bw_usage, 1.f, 5.f);
stop_adaptive_stream(marielle,margaux);
}
}
void upload_bitrate(const char* codec, int payload, int target_bw, int expect_bw) {
bool_t supported = ms_filter_codec_supported(codec);
if( supported ) {
float upload_bw;
stream_manager_t * marielle, * margaux;
start_adaptive_stream(MSAudio, &marielle, &margaux, payload, target_bw*1000, target_bw*1000, 0, 50,0);
//these tests check that encoders stick to the guidelines, so we must use NOT
//the adaptive algorithm which would modify these guidelines
media_stream_enable_adaptive_bitrate_control(&marielle->audio_stream->ms,FALSE);
iterate_adaptive_stream(marielle, margaux, 15000, NULL, 0);
upload_bw=media_stream_get_up_bw(&marielle->audio_stream->ms) / 1000;
CU_ASSERT_IN_RANGE(upload_bw, expect_bw-2, expect_bw+2);
stop_adaptive_stream(marielle,margaux);
}
}
static void qos_analyzer_on_action_suggested(void *user_data, int datac, const char** datav){
reporting_session_report_t *report = (reporting_session_report_t*)user_data;
LinphoneCall *call = report->call;
char * appendbuf;
int i;
int ptime = -1;
int bitrate[3] = {-1, -1, -1};
int up_bw[3] = {-1, -1, -1};
int down_bw[3] = {-1, -1, -1};
MediaStream *streams[3] = { (MediaStream*) call->audiostream, (MediaStream *) call->videostream, (MediaStream *) call->textstream };
for (i = 0; i < 3; i++){
if (streams[i] != NULL){
if (streams[i]->encoder != NULL){
if (ms_filter_has_method(streams[i]->encoder,MS_FILTER_GET_BITRATE)){
ms_filter_call_method(streams[i]->encoder,MS_FILTER_GET_BITRATE,&bitrate[i]);
bitrate[i] /= 1000;
}
}
up_bw[i] = (int)(media_stream_get_up_bw(streams[i])/1000.f);
down_bw[i] = (int)(media_stream_get_down_bw(streams[i])/1000.f);
}
}
if (call->audiostream!=NULL){
if (call->audiostream->ms.encoder!=NULL){
if(ms_filter_has_method(call->audiostream->ms.encoder,MS_AUDIO_ENCODER_GET_PTIME)){
ms_filter_call_method(call->audiostream->ms.encoder,MS_AUDIO_ENCODER_GET_PTIME,&ptime);
}
}
}
appendbuf=ms_strdup_printf("%s%d;", report->qos_analyzer.timestamp?report->qos_analyzer.timestamp:"", ms_time(0));
STR_REASSIGN(report->qos_analyzer.timestamp,appendbuf);
STR_REASSIGN(report->qos_analyzer.input_leg, ms_strdup_printf("%s aenc_ptime aenc_br a_dbw a_ubw venc_br v_dbw v_ubw tenc_br t_dbw t_ubw", datav[0]));
appendbuf=ms_strdup_printf("%s%s %d %d %d %d %d %d %d %d %d %d;", report->qos_analyzer.input?report->qos_analyzer.input:"", datav[1],
ptime, bitrate[0], down_bw[0], up_bw[0], bitrate[1], down_bw[1], up_bw[1], bitrate[2], down_bw[2], up_bw[2]);
STR_REASSIGN(report->qos_analyzer.input,appendbuf);
STR_REASSIGN(report->qos_analyzer.output_leg, ms_strdup(datav[2]));
appendbuf=ms_strdup_printf("%s%s;", report->qos_analyzer.output?report->qos_analyzer.output:"", datav[3]);
STR_REASSIGN(report->qos_analyzer.output, appendbuf);
}
static void audio_stream_dtmf(int codec_payload, int initial_bitrate,int target_bw, int max_recv_rtcp_packet) {
stream_manager_t * marielle = stream_manager_new();
stream_manager_t * margaux = stream_manager_new();
int pause_time=0;
OrtpNetworkSimulatorParams params={0};
params.enabled=TRUE;
params.loss_rate=0;
params.max_bandwidth=target_bw;
params.max_buffer_size=initial_bitrate;
float recv_send_bw_ratio;
int rtcp_interval = 1000;
float marielle_send_bw;
media_stream_enable_adaptive_bitrate_control(&marielle->stream->ms,TRUE);
stream_manager_start(marielle,codec_payload, margaux->local_rtp,initial_bitrate,HELLO_16K_1S_FILE,NULL);
ms_filter_call_method(marielle->stream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
unlink("blibi.wav");
stream_manager_start(margaux,codec_payload, marielle->local_rtp,-1,NULL,"blibi.wav");
rtp_session_enable_network_simulation(margaux->stream->ms.session,¶ms);
rtp_session_set_rtcp_report_interval(margaux->stream->ms.session, rtcp_interval);
wait_for_until(&marielle->stream->ms,&margaux->stream->ms,&marielle->stats.number_of_EndOfFile,10,rtcp_interval*max_recv_rtcp_packet);
marielle_send_bw=media_stream_get_up_bw(&marielle->stream->ms);
recv_send_bw_ratio=params.max_bandwidth/marielle_send_bw;
ms_message("marielle sent bw= [%f] , target was [%f] recv/send [%f]",marielle_send_bw,params.max_bandwidth,recv_send_bw_ratio);
CU_ASSERT_TRUE(recv_send_bw_ratio>0.9);
stream_manager_delete(marielle);
stream_manager_delete(margaux);
}