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echo-cancel: Enable different sample specs for rec and out stream

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Enable advanced AEC methods to use different specs (i.e., number of
channels) for rec and out stream. A typical application is beam forming
resp. multi-channel AEC, which takes multiple record channels to produce
an echo-canceled output stream.
This commit alters the EC API as follows: the EC's init() used to get
source and sink's sample spec/channel map. The new interface renamed
source to rec and sink to play and additionally passes sample spec and
channel map of the out stream. The new parameter names of init()
{rec,play,out}_{ss,map} are more intuitive and also resemble to the
parameter names known from run(). Both rec_{ss,map} and out_{ss,map} are
initialized as we knew it from source_{ss,map} before being passed to
init(). The previous EC implementations only require trivial changes,
i.e., setting rec_{ss,map} to out_{ss,map} at the end of init() in case
that out_{ss,map} is modified in init().
This commit is contained in:
Stefan Huber 2013-02-18 16:31:03 +01:00 • committed by Tanu Kaskinen
parent 17b3cb954b
commit 163430f05b
6 changed files with 132 additions and 94 deletions
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27
src/modules/echo-cancel/webrtc.cc
View file

@ -77,8 +77,9 @@ static int routing_mode_from_string(const char *rmode) {
}
pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
pa_sample_spec *source_ss, pa_channel_map *source_map,
pa_sample_spec *sink_ss, pa_channel_map *sink_map,
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,
uint32_t *nframes, const char *args)
{
webrtc::AudioProcessing *apm = NULL;
@ -158,23 +159,25 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
apm = webrtc::AudioProcessing::Create(0);
source_ss->format = PA_SAMPLE_S16NE;
*sink_ss = *source_ss;
out_ss->format = PA_SAMPLE_S16NE;
*play_ss = *out_ss;
/* FIXME: the implementation actually allows a different number of
* source/sink channels. Do we want to support that? */
*sink_map = *source_map;
*play_map = *out_map;
*rec_ss = *out_ss;
*rec_map = *out_map;
apm->set_sample_rate_hz(source_ss->rate);
apm->set_sample_rate_hz(out_ss->rate);
apm->set_num_channels(source_ss->channels, source_ss->channels);
apm->set_num_reverse_channels(sink_ss->channels);
apm->set_num_channels(out_ss->channels, out_ss->channels);
apm->set_num_reverse_channels(play_ss->channels);
if (hpf)
apm->high_pass_filter()->Enable(true);
if (!mobile) {
if (ec->params.drift_compensation) {
apm->echo_cancellation()->set_device_sample_rate_hz(source_ss->rate);
apm->echo_cancellation()->set_device_sample_rate_hz(out_ss->rate);
apm->echo_cancellation()->enable_drift_compensation(true);
} else {
apm->echo_cancellation()->enable_drift_compensation(false);
@ -215,9 +218,9 @@ pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
apm->voice_detection()->Enable(true);
ec->params.priv.webrtc.apm = apm;
ec->params.priv.webrtc.sample_spec = *source_ss;
ec->params.priv.webrtc.blocksize = (uint64_t)pa_bytes_per_second(source_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
*nframes = ec->params.priv.webrtc.blocksize / pa_frame_size(source_ss);
ec->params.priv.webrtc.sample_spec = *out_ss;
ec->params.priv.webrtc.blocksize = (uint64_t)pa_bytes_per_second(out_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
*nframes = ec->params.priv.webrtc.blocksize / pa_frame_size(out_ss);
pa_modargs_free(ma);
return TRUE;