--- a/help/ht0-buildagraph.dox
+++ b/help/ht0-buildagraph.dox
@@ -114,7 +114,7 @@ cards' filters:
Then you need to 'attach' the filters to a ticker. A ticker is a graph
manager responsible for running filters.
-In the above case, there is 2 independant graph within the ticker: you
+In the above case, there are two (2) independent graphs within the ticker: you
need to attach the first element of each graph (the one that does not
contains any INPUT pins)
--- a/include/mediastreamer2/msfileplayer.h
+++ b/include/mediastreamer2/msfileplayer.h
@@ -30,7 +30,7 @@
/* set loop mode:
-1: no looping,
0: loop at end of file,
- x>0, loop after x miliseconds after eof
+ x>0, loop after x milliseconds after EOF
*/
#define MS_FILE_PLAYER_LOOP MS_FILTER_METHOD(MS_FILE_PLAYER_ID,4,int)
#define MS_FILE_PLAYER_DONE MS_FILTER_METHOD(MS_FILE_PLAYER_ID,5,int)
--- a/include/mediastreamer2/msticker.h
+++ b/include/mediastreamer2/msticker.h
@@ -39,7 +39,7 @@
/**
- * Function pointer for method getting time in miliseconds from an external source.
+ * Function pointer for method of getting time in milliseconds from an external source.
* @var MSTickerTimeFunc
*/
typedef uint64_t (*MSTickerTimeFunc)(void *);
@@ -77,7 +77,7 @@ struct _MSTicker
MSList *execution_list; /* the list of source filters to be executed.*/
MSList *task_list; /* list of tasks (see ms_filter_postpone_task())*/
ms_thread_t thread; /* the thread ressource*/
- int interval; /* in miliseconds*/
+ int interval; /* in milliseconds*/
int exec_id;
uint32_t ticks;
uint64_t time; /* a time since the start of the ticker expressed in milisec*/
--- a/src/audiofilters/genericplc.h
+++ b/src/audiofilters/genericplc.h
@@ -26,7 +26,7 @@
/* 2/<min frequency we want to be able to reproduce> gives then length in seconds */
#define PLC_BUFFER_LEN 2/40
-/* define in ms the maximum duration of PLC(after wich the output will be 0), and after how long we start decreasing the output volume to reach 0 at MAX_PLC_LEN */
+/* define in ms the maximum duration of PLC(after which the output will be 0), and how long before we start decreasing the output volume to reach 0 at MAX_PLC_LEN */
#define PLC_DECREASE_START 100
#define MAX_PLC_LEN 150
--- a/src/base/msticker.c
+++ b/src/base/msticker.c
@@ -454,7 +454,7 @@ void * ms_ticker_run(void *arg)
s->time+=s->interval;
late=s->wait_next_tick(s->wait_next_tick_data,s->time);
if (late>s->interval*5 && late>lastlate){
- ms_warning("%s: We are late of %d miliseconds.",s->name,late);
+ ms_warning("%s: We are late by %d milliseconds.",s->name,late);
late_tick_time=ms_get_cur_time_ms();
}
lastlate=late;
--- a/src/crypto/dtls_srtp.c
+++ b/src/crypto/dtls_srtp.c
@@ -231,7 +231,7 @@ static void schedule_rtcp(struct _RtpTra
}
/**
* Check if the incoming message is a DTLS packet.
- * If it is, store it in the context incoming buffer and call the polarssl function wich will process it.
+ * If it is, store it in the context incoming buffer and call the polarssl function which will process it.
* This function also manages the client retransmission timer
*
* @param[in] msg the incoming message
@@ -346,7 +346,7 @@ static bool_t ms_dtls_srtp_process_dtls_
base_index += Handshake_Header_Length + frag_length; // bytes parsed so far
frag += Handshake_Header_Length + frag_length; // point to the begining of the next fragment
} else { // message is malformed in a nasty way
- ms_warning("DTLS Received %s packet len %d sessions: %p rtp session %p is malformed in an agressive way", is_rtp==TRUE?"RTP":"RTCP", (int)msgLength, ctx->stream_sessions, ctx->stream_sessions->rtp_session);
+ ms_warning("DTLS Received %s packet len %d sessions: %p rtp session %p is malformed in an aggressive way", is_rtp==TRUE?"RTP":"RTCP", (int)msgLength, ctx->stream_sessions, ctx->stream_sessions->rtp_session);
base_index = msgLength; // get out of the while
ms_free(reassembled_packet);
reassembled_packet = NULL;
@@ -644,7 +644,7 @@ static int ms_dtls_srtp_rtp_process_on_r
}
if (ctx->role != MSDtlsSrtpRoleIsServer) { /* close the connection only if we are client, if we are server, the client may ask again for last packets */
- /*FireFox version 43 requires DTLS channel to be kept openned, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtp_dtls_context->ssl) );*/
+ /*FireFox version 43 requires DTLS channel to be kept open, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtp_dtls_context->ssl) );*/
}
@@ -725,7 +725,7 @@ static int ms_dtls_srtp_rtcp_process_on_
}
if (ctx->role != MSDtlsSrtpRoleIsServer) { /* close the connection only if we are client, if we are server, the client may ask again for last packets */
- /*FireFox version 43 requires DTLS channel to be kept openned, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtcp_dtls_context->ssl) );*/
+ /*FireFox version 43 requires DTLS channel to be kept open, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtcp_dtls_context->ssl) );*/
}
}
@@ -845,7 +845,7 @@ void ms_dtls_srtp_set_peer_fingerprint(M
size_t peer_fingerprint_length = strlen(peer_fingerprint)+1; // include the null termination
if (peer_fingerprint_length>sizeof(context->peer_fingerprint)) {
memcpy(context->peer_fingerprint, peer_fingerprint, sizeof(context->peer_fingerprint));
- ms_error("DTLS-SRTP received from SDP INVITE a peer fingerprint %d bytes length wich is longer than maximum storage %d bytes", (int)peer_fingerprint_length, (int)sizeof(context->peer_fingerprint));
+ ms_error("DTLS-SRTP received from SDP INVITE a peer fingerprint %d bytes long, which is longer than maximum storage of %d bytes", (int)peer_fingerprint_length, (int)sizeof(context->peer_fingerprint));
} else {
memcpy(context->peer_fingerprint, peer_fingerprint, peer_fingerprint_length);
}
--- a/src/crypto/ms_srtp.c
+++ b/src/crypto/ms_srtp.c
@@ -589,6 +589,6 @@ const char * ms_srtp_stream_type_to_stri
case MSSRTP_RTCP_STREAM: return "MSSRTP_RTCP_STREAM";
case MSSRTP_ALL_STREAMS: return "MSSRTP_ALL_STREAMS";
}
- return "Unkown srtp tream type";
+ return "Unknown srtp stream type";
}
--- a/src/crypto/zrtp.c
+++ b/src/crypto/zrtp.c
@@ -641,7 +641,7 @@ MSZrtpContext* ms_zrtp_multistream_new(M
int retval;
MSZrtpContext *userData;
if ((retval = bzrtp_addChannel(activeContext->zrtpContext, sessions->rtp_session->snd.ssrc)) != 0) {
- ms_warning("ZRTP could't add stream, returns %x", retval);
+ ms_warning("ZRTP couldn't add stream, returned %x", retval);
}
ms_message("Initializing multistream ZRTP context on rtp session [%p] ssrc 0x%x",sessions->rtp_session, sessions->rtp_session->snd.ssrc);
--- a/src/otherfilters/msrtp.c
+++ b/src/otherfilters/msrtp.c
@@ -731,7 +731,7 @@ static void receiver_process(MSFilter *
return;
if (d->reset_jb){
- ms_message("Reseting jitter buffer");
+ ms_message("Resetting jitter buffer");
rtp_session_resync(d->session);
d->reset_jb=FALSE;
}
--- a/src/utils/audiodiff.c
+++ b/src/utils/audiodiff.c
@@ -283,7 +283,7 @@ static int _ms_audio_diff_chunked(FileIn
*ret = cum_res / (double)tot_energy;
ms_message("Similarity factor weighted with most significant chunks is [%g]", *ret);
*ret = *ret * (1-variance);
- ms_message("After integrating max position variance accross chunks, it is [%g]", *ret);
+ ms_message("After integrating maximum position variance across chunks, it is [%g]", *ret);
ms_free(chunk_energies);
ms_free(max_pos_table);
return maxpos;
--- a/src/utils/mkv_reader.h
+++ b/src/utils/mkv_reader.h
@@ -136,7 +136,7 @@ MKVTrackReader *mkv_reader_get_track_rea
/**
* @brief Set the reading head of each assocated track reader at a specific position
* @param reader MKVReader
- * @param pos_ms Position of the head in miliseconds
+ * @param pos_ms Position of the head in milliseconds
* @return The effective position of the head after the operation
*/
int mkv_reader_seek(MKVReader *reader, int pos_ms);
--- a/src/videofilters/bb10_capture.cpp
+++ b/src/videofilters/bb10_capture.cpp
@@ -125,7 +125,7 @@ static void bb10capture_open_camera(BB10
camera_error_t error;
if (d->camera_openned) {
- ms_warning("[bb10_capture] camera already openned, skipping...");
+ ms_warning("[bb10_capture] camera already opened, skipping...");
return;
}
@@ -157,7 +157,7 @@ static void bb10capture_open_camera(BB10
static void bb10capture_start_capture(BB10Capture *d) {
if (!d->camera_openned) {
- ms_error("[bb10_capture] camera not openned, skipping...");
+ ms_error("[bb10_capture] camera not opened, skipping...");
return;
}
if (d->capture_started) {
@@ -186,7 +186,7 @@ static void bb10capture_stop_capture(BB1
static void bb10capture_close_camera(BB10Capture *d) {
if (!d->camera_openned) {
- ms_warning("[bb10_capture] camera not openned, skipping...");
+ ms_warning("[bb10_capture] camera not opened, skipping...");
return;
}
--- a/src/videofilters/msv4l2.c
+++ b/src/videofilters/msv4l2.c
@@ -697,7 +697,7 @@ static void *msv4l2_thread(void *ptr){
ms_message("msv4l2_thread starting");
if (s->fd==-1){
if( msv4l2_open(s)!=0){
- ms_warning("msv4l2 could not be openned");
+ ms_warning("msv4l2 could not be opened");
goto close;
}
}
--- a/src/videofilters/vp8.c
+++ b/src/videofilters/vp8.c
@@ -407,10 +407,10 @@ static void enc_fill_encoder_flags(EncSt
} else if (frame_type & VP8_ALTR_FRAME) {
*flags |= (VP8_EFLAG_FORCE_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_REF_ARF);
if (s->frame_count > s->frames_state.last_independent_frame + 5*enc_get_ref_frames_interval(s)){
- /*force an independant alt ref frame to force picture to be refreshed completely, otherwise
+ /*force an independent alt ref frame to force picture to be refreshed completely, otherwise
* pixel color saturation appears due to accumulation of small predictive errors*/
*flags |= VP8_EFLAG_NO_REF_LAST | VP8_EFLAG_NO_REF_GF;
- ms_message("Forcing independant altref frame.");
+ ms_message("Forcing independent altref frame.");
}
}
if (!(*flags & VPX_EFLAG_FORCE_KF)){
@@ -1141,7 +1141,7 @@ static int dec_freeze_on_error(MSFilter
static int dec_reset(MSFilter *f, void *data) {
DecState *s = (DecState *)f->data;
- ms_message("Reseting VP8 decoder");
+ ms_message("Resetting VP8 decoder");
ms_filter_lock(f);
vpx_codec_destroy(&s->codec);
if (dec_initialize_impl(f) != 0){
--- a/src/voip/audiostream.c
+++ b/src/voip/audiostream.c
@@ -914,7 +914,7 @@ int audio_stream_start_from_io(AudioStre
}
/* sample rate is already set for rtpsend and rtprcv, check if we have to adjust it to */
- /* be able to use the echo canceller wich may be limited (webrtc aecm max frequency is 16000 Hz) */
+ /* be able to use the echo canceller, which may be limited (webrtc aecm max frequency is 16000 Hz) */
// First check if we need to use the echo canceller
// Overide feature if not requested or done at sound card level
if ( ((stream->features & AUDIO_STREAM_FEATURE_EC) && !stream->use_ec) || has_builtin_ec )
--- a/src/voip/msvideo.c
+++ b/src/voip/msvideo.c
@@ -955,7 +955,7 @@ void ms_average_fps_init(MSAverageFPS* a
afps->mean_inter_frame = 0;
afps->context = ctx;
if (!ctx || strstr(ctx, "%f") == 0) {
- ms_error("Invalid MSAverageFPS context given '%s' (must be not null and must contain one occurence of '%%f'", ctx);
+ ms_error("Invalid MSAverageFPS context given '%s' (must be not null and must contain one occurrence of '%%f'", ctx);
}
}
--- a/tools/mediastream.c
+++ b/tools/mediastream.c
@@ -198,7 +198,7 @@ const char *usage="mediastream --local <
"[ --ec-tail <echo canceller tail length in ms> ]\n"
"[ --el (enable echo limiter) ]\n"
"[ --el-force <(float) [0-1]> (The proportional coefficient controlling the mic attenuation) ]\n"
- "[ --el-speed <(float) [0-1]> (gain changes are smoothed with a coefficent) ]\n"
+ "[ --el-speed <(float) [0-1]> (gain changes are smoothed with a coefficient) ]\n"
"[ --el-sustain <(int)> (Time in milliseconds for which the attenuation is kept unchanged after) ]\n"
"[ --el-thres <(float) [0-1]> (Threshold above which the system becomes active) ]\n"
"[ --el-transmit-thres <(float) [0-1]> (TO BE DOCUMENTED) ]\n"
@@ -210,7 +210,7 @@ const char *usage="mediastream --local <
"[ --ice-remote-candidate <ip:port:[host|srflx|prflx|relay]> ]\n"
"[ --infile <input wav file> specify a wav file to be used for input, instead of soundcard ]\n"
"[ --interactive (run in interactive mode) ]\n"
- "[ --jitter <miliseconds> ]\n"
+ "[ --jitter <milliseconds> ]\n"
"[ --log <file> ]\n"
"[ --mtu <mtu> (specify MTU)]\n"
"[ --netsim-bandwidth <bandwidth limit in bits/s> (simulates a network download bandwidth limit) ]\n"
@@ -228,7 +228,7 @@ const char *usage="mediastream --local <
"[ --outfile <output wav file> specify a wav file to write audio into, instead of soundcard ]\n"
"[ --playback-card <name> ]\n"
"[ --rc <rate control algorithm> possible values are: none, simple, advanced ]\n"
- "[ --srtp <local master_key> <remote master_key> (enable srtp, master key is generated if absent from comand line) ]\n"
+ "[ --srtp <local master_key> <remote master_key> (enable srtp, master key is generated if absent from command line) ]\n"
"[ --verbose (most verbose messages) ]\n"
"[ --video-display-filter <name> ]\n"
"[ --video-windows-id <video surface:preview surface >]\n"
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