diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h index ab0c5e91e..83f8b8534 100644 --- a/src/modules/echo-cancel/echo-cancel.h +++ b/src/modules/echo-cancel/echo-cancel.h @@ -66,6 +66,7 @@ struct pa_echo_canceller_params { void *apm; unsigned int blocksize; /* in frames */ pa_sample_spec rec_ss, play_ss, out_ss; + float *rec_buffer[PA_CHANNELS_MAX], *play_buffer[PA_CHANNELS_MAX]; /* for deinterleaved buffers */ void *trace_callback; bool agc; bool first; diff --git a/src/modules/echo-cancel/module-echo-cancel.c b/src/modules/echo-cancel/module-echo-cancel.c index 484281f01..4e898d070 100644 --- a/src/modules/echo-cancel/module-echo-cancel.c +++ b/src/modules/echo-cancel/module-echo-cancel.c @@ -2158,12 +2158,12 @@ int main(int argc, char* argv[]) { goto fail; } - source_ss.format = PA_SAMPLE_S16LE; + source_ss.format = PA_SAMPLE_FLOAT32LE; source_ss.rate = DEFAULT_RATE; source_ss.channels = DEFAULT_CHANNELS; pa_channel_map_init_auto(&source_map, source_ss.channels, PA_CHANNEL_MAP_DEFAULT); - sink_ss.format = PA_SAMPLE_S16LE; + sink_ss.format = PA_SAMPLE_FLOAT32LE; sink_ss.rate = DEFAULT_RATE; sink_ss.channels = DEFAULT_CHANNELS; pa_channel_map_init_auto(&sink_map, sink_ss.channels, PA_CHANNEL_MAP_DEFAULT); diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc index 6c2dd99f1..93af18f76 100644 --- a/src/modules/echo-cancel/webrtc.cc +++ b/src/modules/echo-cancel/webrtc.cc @@ -117,8 +117,8 @@ static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec 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_S16NE; - play_ss->format = PA_SAMPLE_S16NE; + rec_ss->format = PA_SAMPLE_FLOAT32NE; + play_ss->format = PA_SAMPLE_FLOAT32NE; /* AudioProcessing expects one of the following rates */ if (rec_ss->rate >= 48000) @@ -130,7 +130,6 @@ static void pa_webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec else rec_ss->rate = 8000; - /* In int16 mode, AudioProcessing will give us the same spec we give it */ *out_ss = *rec_ss; *out_map = *rec_map; @@ -147,7 +146,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec, webrtc::ProcessingConfig pconfig; webrtc::Config config; bool hpf, ns, agc, dgc, mobile, cn, vad, ext_filter, intelligibility, experimental_agc; - int rm = -1; + int rm = -1, i; uint32_t agc_start_volume; pa_modargs *ma; bool trace = false; @@ -340,6 +339,11 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec, *nframes = ec->params.webrtc.blocksize; ec->params.webrtc.first = true; + for (i = 0; i < rec_ss->channels; i++) + ec->params.webrtc.rec_buffer[i] = pa_xnew(float, *nframes); + for (i = 0; i < play_ss->channels; i++) + ec->params.webrtc.play_buffer[i] = pa_xnew(float, *nframes); + pa_modargs_free(ma); return true; @@ -357,18 +361,14 @@ fail: void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) { webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; - webrtc::AudioFrame play_frame; const pa_sample_spec *ss = &ec->params.webrtc.play_ss; + int n = ec->params.webrtc.blocksize; + float **buf = ec->params.webrtc.play_buffer; + webrtc::StreamConfig config(ss->rate, ss->channels, false); - play_frame.num_channels_ = ss->channels; - play_frame.sample_rate_hz_ = ss->rate; - play_frame.interleaved_ = true; - play_frame.samples_per_channel_ = ec->params.webrtc.blocksize; + pa_deinterleave(play, (void **) buf, ss->channels, pa_sample_size(ss), n); - pa_assert(play_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples); - memcpy(play_frame.data_, play, ec->params.webrtc.blocksize * pa_frame_size(ss)); - - pa_assert_se(apm->ProcessReverseStream(&play_frame) == webrtc::AudioProcessing::kNoError); + pa_assert_se(apm->ProcessReverseStream(buf, config, config, buf) == webrtc::AudioProcessing::kNoError); /* FIXME: If ProcessReverseStream() makes any changes to the audio, such as * applying intelligibility enhancement, those changes don't have any @@ -379,19 +379,16 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) { void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) { webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; - webrtc::AudioFrame out_frame; const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss; const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss; + float **buf = ec->params.webrtc.rec_buffer; + int n = ec->params.webrtc.blocksize; pa_cvolume v; int old_volume, new_volume; + webrtc::StreamConfig rec_config(rec_ss->rate, rec_ss->channels, false); + webrtc::StreamConfig out_config(out_ss->rate, out_ss->channels, false); - out_frame.num_channels_ = rec_ss->channels; - out_frame.sample_rate_hz_ = rec_ss->rate; - out_frame.interleaved_ = true; - out_frame.samples_per_channel_ = ec->params.webrtc.blocksize; - - pa_assert(out_frame.samples_per_channel_ <= webrtc::AudioFrame::kMaxDataSizeSamples); - memcpy(out_frame.data_, rec, ec->params.webrtc.blocksize * pa_frame_size(rec_ss)); + pa_deinterleave(rec, (void **) buf, rec_ss->channels, pa_sample_size(rec_ss), n); if (ec->params.webrtc.agc) { pa_cvolume_init(&v); @@ -401,7 +398,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out } apm->set_stream_delay_ms(0); - pa_assert_se(apm->ProcessStream(&out_frame) == webrtc::AudioProcessing::kNoError); + pa_assert_se(apm->ProcessStream(buf, rec_config, out_config, buf) == webrtc::AudioProcessing::kNoError); if (ec->params.webrtc.agc) { if (PA_UNLIKELY(ec->params.webrtc.first)) { @@ -421,7 +418,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out } } - memcpy(out, out_frame.data_, ec->params.webrtc.blocksize * pa_frame_size(out_ss)); + pa_interleave((const void **) buf, out_ss->channels, out, pa_sample_size(out_ss), n); } void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) { @@ -436,6 +433,8 @@ void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t * } void pa_webrtc_ec_done(pa_echo_canceller *ec) { + int i; + if (ec->params.webrtc.trace_callback) { webrtc::Trace::ReturnTrace(); delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback); @@ -445,4 +444,9 @@ void pa_webrtc_ec_done(pa_echo_canceller *ec) { delete (webrtc::AudioProcessing*)ec->params.webrtc.apm; ec->params.webrtc.apm = NULL; } + + for (i = 0; i < ec->params.webrtc.rec_ss.channels; i++) + pa_xfree(ec->params.webrtc.rec_buffer[i]); + for (i = 0; i < ec->params.webrtc.play_ss.channels; i++) + pa_xfree(ec->params.webrtc.play_buffer[i]); }