Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Misspellings in source code (patch file included) #24

Open
RogueScholar opened this issue Jul 16, 2020 · 1 comment
Open

Misspellings in source code (patch file included) #24

RogueScholar opened this issue Jul 16, 2020 · 1 comment

Comments

@RogueScholar
Copy link

While working on packaging version 4.4.0 for Debian/Ubuntu I encountered some misspelled words in the code comments and took a few extra minutes to craft a patch file to correct them. I'm attaching it to this issue in case you'd like to apply it here upstream.

Patch file:
mediastreamer2-corrected-misspellings.txt

Patch file contents
--- 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"

I've downloaded and filled out your Contributor Agreement as well, in case you need it on file to use the patch. If so, just let me know which method you'd prefer I use to transmit it to you and it's yours.

Warmly,
Peter

@Viish
Copy link
Member

Viish commented Jul 17, 2020

Hi,
Yes, we will be very glad to merge it.
Can you send me your contributor agreement by email?
Thanks a lot!

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment
Labels
None yet
Projects
None yet
Development

No branches or pull requests

2 participants