[pulseaudio-discuss] [PATCH v3 17/24] echo-cancel: Fix webrtc canceller when rec channels != play channels

[Arun Raghavan]

On 8 February 2016 at 13:32, Tanu Kaskinen <tanuk at iki.fi> wrote:
> On Mon, 2016-01-18 at 13:06 +0530, arun at accosted.net wrote:
>> From: Arun Raghavan <git at arunraghavan.net>
>>
>> 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