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| author | Arun Raghavan <git@arunraghavan.net> | 2016-02-17 19:47:14 +0530 |
|---|---|---|
| committer | Arun Raghavan <git@arunraghavan.net> | 2016-02-25 09:17:25 +0530 |
| commit | cfd3a948deb4d2fc0f2a2d3a5de7f466311b6478 (patch) | |
| tree | 2a4b3889172613cbcc139c534058138351aab7dc | |
| parent | be5e196c1a6065529e44a49e78ca75050706e321 (diff) | |
| download | pulseaudio-cfd3a948deb4d2fc0f2a2d3a5de7f466311b6478.tar.gz | |
echo-cancel: Remove pa_ prefix on private functions
| -rw-r--r-- | src/modules/echo-cancel/speex.c | 8 | ||||
| -rw-r--r-- | src/modules/echo-cancel/webrtc.cc | 8 |
2 files changed, 8 insertions, 8 deletions
diff --git a/src/modules/echo-cancel/speex.c b/src/modules/echo-cancel/speex.c index 08c1027b3..a3ae64673 100644 --- a/src/modules/echo-cancel/speex.c +++ b/src/modules/echo-cancel/speex.c @@ -47,9 +47,9 @@ static const char* const valid_modargs[] = { NULL }; -static void pa_speex_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map, - pa_sample_spec *play_ss, pa_channel_map *play_map, - pa_sample_spec *out_ss, pa_channel_map *out_map) { +static void speex_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map, + pa_sample_spec *play_ss, pa_channel_map *play_map, + pa_sample_spec *out_ss, pa_channel_map *out_map) { out_ss->format = PA_SAMPLE_S16NE; *play_ss = *out_ss; @@ -170,7 +170,7 @@ bool pa_speex_ec_init(pa_core *c, pa_echo_canceller *ec, goto fail; } - pa_speex_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map); + speex_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map); rate = out_ss->rate; *nframes = pa_echo_canceller_blocksize_power2(rate, frame_size_ms); diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc index 756d5f7a0..73aeda218 100644 --- a/src/modules/echo-cancel/webrtc.cc +++ b/src/modules/echo-cancel/webrtc.cc @@ -114,9 +114,9 @@ static pa_volume_t webrtc_volume_to_pa(int v) return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME; } -static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map, - pa_sample_spec *play_ss, pa_channel_map *play_map, - pa_sample_spec *out_ss, pa_channel_map *out_map) +static void webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map, + pa_sample_spec *play_ss, pa_channel_map *play_map, + pa_sample_spec *out_ss, pa_channel_map *out_map) { rec_ss->format = PA_SAMPLE_FLOAT32NE; play_ss->format = PA_SAMPLE_FLOAT32NE; @@ -276,7 +276,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec, webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback); } - pa_webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map); + webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map); apm = webrtc::AudioProcessing::Create(config); |