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https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git
synced 2025-11-02 09:01:46 -05:00
echo-cancel: webrtc canceller supports different in/out channel counts
Needed for upcoming beamforming code.
This commit is contained in:
parent
222a98846c
commit
3133ff8e11
2 changed files with 9 additions and 7 deletions
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@ -65,7 +65,7 @@ struct pa_echo_canceller_params {
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* to C++ linkage. apm is a pointer to an AudioProcessing object */
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void *apm;
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unsigned int blocksize; /* in frames */
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pa_sample_spec rec_ss, play_ss;
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pa_sample_spec rec_ss, play_ss, out_ss;
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void *trace_callback;
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bool agc;
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bool first;
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@ -335,6 +335,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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ec->params.webrtc.apm = apm;
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ec->params.webrtc.rec_ss = *rec_ss;
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ec->params.webrtc.play_ss = *play_ss;
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ec->params.webrtc.out_ss = *out_ss;
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ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC;
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*nframes = ec->params.webrtc.blocksize;
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ec->params.webrtc.first = true;
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@ -379,17 +380,18 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
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void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
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webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
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webrtc::AudioFrame out_frame;
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const pa_sample_spec *ss = &ec->params.webrtc.rec_ss;
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const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss;
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const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss;
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pa_cvolume v;
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int old_volume, new_volume;
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out_frame.num_channels_ = ss->channels;
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out_frame.sample_rate_hz_ = ss->rate;
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out_frame.num_channels_ = rec_ss->channels;
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out_frame.sample_rate_hz_ = rec_ss->rate;
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out_frame.interleaved_ = true;
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out_frame.samples_per_channel_ = ec->params.webrtc.blocksize;
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pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples);
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memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(ss));
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memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(rec_ss));
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if (ec->params.webrtc.agc) {
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pa_cvolume_init(&v);
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@ -414,12 +416,12 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
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}
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if (old_volume != new_volume) {
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pa_cvolume_set(&v, ss->channels, webrtc_volume_to_pa(new_volume));
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pa_cvolume_set(&v, rec_ss->channels, webrtc_volume_to_pa(new_volume));
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pa_echo_canceller_set_capture_volume(ec, &v);
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}
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}
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memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(ss));
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memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(out_ss));
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}
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void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
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