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| author | Aaron Boxer <aaron.boxer@collabora.com> | 2019-09-02 15:08:44 -0400 |
|---|---|---|
| committer | Aaron Boxer <aaron.boxer@collabora.com> | 2019-11-05 09:11:25 -0500 |
| commit | 6d3429af34ed0b5905faf32d2f22b9db2451f116 (patch) | |
| tree | f18e8fed7a80ddd4db89a76bab109fcd37a76819 /ext/srtp | |
| parent | 2386858a9179aff2ec249bdffa904bf407de455f (diff) | |
| download | gstreamer-plugins-bad-6d3429af34ed0b5905faf32d2f22b9db2451f116.tar.gz | |
documentation: fixed a heap o' typos
Diffstat (limited to 'ext/srtp')
| -rw-r--r-- | ext/srtp/gstsrtpdec.c | 8 | ||||
| -rw-r--r-- | ext/srtp/gstsrtpenc.c | 6 |
2 files changed, 7 insertions, 7 deletions
diff --git a/ext/srtp/gstsrtpdec.c b/ext/srtp/gstsrtpdec.c index 47ebfae0b..33880bc9e 100644 --- a/ext/srtp/gstsrtpdec.c +++ b/ext/srtp/gstsrtpdec.c @@ -292,7 +292,7 @@ gst_srtp_dec_class_init (GstSrtpDecClass * klass) * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * - * Signal emited to get the parameters relevant to stream + * Signal emitted to get the parameters relevant to stream * with @ssrc. User should provide the key and the RTP and * RTCP encryption ciphers and authentication, and return * them wrapped in a GstCaps. @@ -318,7 +318,7 @@ gst_srtp_dec_class_init (GstSrtpDecClass * klass) * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * - * Signal emited when the stream with @ssrc has reached the + * Signal emitted when the stream with @ssrc has reached the * soft limit of utilisation of it's master encryption key. * User should provide a new key and new RTP and RTCP encryption * ciphers and authentication, and return them wrapped in a @@ -333,7 +333,7 @@ gst_srtp_dec_class_init (GstSrtpDecClass * klass) * @gstsrtpdec: the element on which the signal is emitted * @ssrc: The unique SSRC of the stream * - * Signal emited when the stream with @ssrc has reached the + * Signal emitted when the stream with @ssrc has reached the * hard limit of utilisation of it's master encryption key. * User should provide a new key and new RTP and RTCP encryption * ciphers and authentication, and return them wrapped in a @@ -361,7 +361,7 @@ gst_srtp_dec_class_init (GstSrtpDecClass * klass) /* initialize the new element * instantiate pads and add them to element - * set pad calback functions + * set pad callback functions * initialize instance structure */ static void diff --git a/ext/srtp/gstsrtpenc.c b/ext/srtp/gstsrtpenc.c index ae6b450ab..d677afcce 100644 --- a/ext/srtp/gstsrtpenc.c +++ b/ext/srtp/gstsrtpenc.c @@ -56,7 +56,7 @@ * An application can request multiple RTP and RTCP pads to protect, * but every sink pad requested must receive packets from the same * source (identical SSRC). If a packet received contains a different - * SSRC, a warning is emited and the valid SSRC is forced on the packet. + * SSRC, a warning is emitted and the valid SSRC is forced on the packet. * * This element uses libsrtp library. When receiving the first packet, * the library is initialized with a new stream (based on the SSRC). It @@ -335,7 +335,7 @@ gst_srtp_enc_class_init (GstSrtpEncClass * klass) * GstSrtpEnc::soft-limit: * @gstsrtpenc: the element on which the signal is emitted * - * Signal emited when the stream with @ssrc has reached the soft + * Signal emitted when the stream with @ssrc has reached the soft * limit of utilisation of it's master encryption key. User should * provide a new key by setting the #GstSrtpEnc:key property. */ @@ -484,7 +484,7 @@ done: return ret; } -/* Release ressources and set default values +/* Release resources and set default values */ static void gst_srtp_enc_reset_no_lock (GstSrtpEnc * filter) |