On 8 February 2016 at 13:32, Tanu Kaskinen <[email protected]> wrote: > On Mon, 2016-01-18 at 13:06 +0530, [email protected] wrote: >> From: Arun Raghavan <[email protected]> >> >> The calculations around how many samples were sent to the canceller >> engine was not updated when we started supporting different channel >> counts for playback and capture. >> --- >> src/modules/echo-cancel/echo-cancel.h | 4 ++-- >> src/modules/echo-cancel/webrtc.cc | 25 +++++++++++++------------ >> 2 files changed, 15 insertions(+), 14 deletions(-) >> >> diff --git a/src/modules/echo-cancel/echo-cancel.h >> b/src/modules/echo-cancel/echo-cancel.h >> index 37f99c0..4693516 100644 >> --- a/src/modules/echo-cancel/echo-cancel.h >> +++ b/src/modules/echo-cancel/echo-cancel.h >> @@ -64,8 +64,8 @@ struct pa_echo_canceller_params { >> /* This is a void* so that we don't have to convert this whole >> file >> * to C++ linkage. apm is a pointer to an AudioProcessing >> object */ >> void *apm; >> - uint32_t blocksize; >> - pa_sample_spec sample_spec; >> + int32_t blocksize; /* in frames */ > > Why is the type changed from unsigned to signed? It doesn't look like > you need negative values.
I meant to change that to just an int, rather than specify the size, which is unnecessary here. >> + pa_sample_spec rec_ss, play_ss; >> bool agc; >> bool trace; >> bool first; >> diff --git a/src/modules/echo-cancel/webrtc.cc >> b/src/modules/echo-cancel/webrtc.cc >> index ec0a383..2732b38 100644 >> --- a/src/modules/echo-cancel/webrtc.cc >> +++ b/src/modules/echo-cancel/webrtc.cc >> @@ -327,9 +327,11 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller >> *ec, >> apm->voice_detection()->Enable(true); >> >> ec->params.webrtc.apm = apm; >> - ec->params.webrtc.sample_spec = *out_ss; >> - ec->params.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * >> BLOCK_SIZE_US / PA_USEC_PER_SEC; >> - *nframes = ec->params.webrtc.blocksize / pa_frame_size(out_ss); >> + ec->params.webrtc.rec_ss = *rec_ss; >> + ec->params.webrtc.play_ss = *play_ss; >> + ec->params.webrtc.blocksize = >> + (uint64_t) (pa_bytes_per_second(out_ss) / pa_frame_size(out_ss)) * >> BLOCK_SIZE_US / PA_USEC_PER_SEC; > > pa_bytes_per_second(out_ss) / pa_frame_size(out_ss) calculates the > sample rate, so it can be replaced with out_ss->rate. Ack. -- Arun _______________________________________________ pulseaudio-discuss mailing list [email protected] https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
