--- /dev/null
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2021 Jaechul Lee <jcsing.lee@samsung.com>
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/cdecl.h>
+
+#include <webrtc/modules/audio_processing/include/audio_processing.h>
+
+#define DEFAULT_PROCESS_SIZE_MS 10
+
+using namespace webrtc;
+
+PA_C_DECL_BEGIN
+#include <pulsecore/log.h>
+#include <pulsecore/macro.h>
+#include <pulsecore/sconv.h>
+#include <pulsecore/sample-util.h>
+#include <pulse/xmalloc.h>
+#include <pulse/sample.h>
+#include <pulse/timeval.h>
+#include <stdint.h>
+
+static void allocate_stream_buffer(struct algo_webrtc *webrtc, size_t nframes);
+static void deallocate_stream_buffer(struct algo_webrtc *webrtc);
+static void convert_s16_to_float(float *dst, int16_t *src, size_t n);
+static void convert_float_to_s16(int16_t *dst, float *src, size_t n);
+
+void *webrtc_audio_create(size_t nframes, pa_sample_spec *ss);
+int32_t webrtc_audio_process(void *priv, int8_t *rec, int8_t *ref, int8_t *out);
+int32_t webrtc_audio_destroy(void *priv);
+PA_C_DECL_END
+
+struct algo_webrtc {
+ AudioProcessing *ap;
+ Config config;
+ StreamConfig *sconfig;
+
+ float *rec_fbuf;
+ float *rec_dbuf[PA_CHANNELS_MAX];
+
+ float *ref_fbuf;
+ float *ref_dbuf[PA_CHANNELS_MAX];
+
+ float *out_fbuf;
+ float *out_dbuf[PA_CHANNELS_MAX];
+
+ pa_sample_spec ss;
+ size_t frames;
+
+ /* Currently, webrtc uses fixed size(10ms) buffer */
+ int loops;
+ size_t fixed_bytes;
+ size_t fixed_frames;
+};
+
+void *webrtc_audio_create(size_t nframes, pa_sample_spec *ss) {
+ struct algo_webrtc *webrtc = NULL;
+ size_t fixed_bytes, request_bytes;
+ Config config;
+
+ pa_assert(ss);
+
+ if (ss->channels > 2 || ss->format != PA_SAMPLE_S16LE) {
+ pa_log_error("Invalid channels (%d) or format (%d)", ss->channels, ss->format);
+ return NULL;
+ }
+
+ fixed_bytes = pa_usec_to_bytes(DEFAULT_PROCESS_SIZE_MS * PA_USEC_PER_MSEC, ss);
+ request_bytes = nframes * pa_frame_size(ss);
+
+ if (fixed_bytes > request_bytes) {
+ pa_log_error("nframes should be bigger than %dms. nframes(%zu) request_bytes(%zu)",
+ DEFAULT_PROCESS_SIZE_MS, nframes, request_bytes);
+ return NULL;
+ }
+
+ if (request_bytes % fixed_bytes) {
+ pa_log_error("request_bytes(%zu) should be multiples of fixed_bytes(%zu)",
+ nframes, request_bytes);
+ return NULL;
+ }
+
+ webrtc = pa_xnew0(struct algo_webrtc, 1);
+ webrtc->ss = *ss;
+ webrtc->fixed_bytes = fixed_bytes;
+ webrtc->fixed_frames = fixed_bytes / pa_frame_size(ss);
+ webrtc->loops = request_bytes / fixed_bytes;
+
+ config.Set<ExperimentalNs>(new ExperimentalNs(false));
+ config.Set<Intelligibility>(new Intelligibility(false));
+ config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
+
+ webrtc->ap = AudioProcessing::Create(config);
+ if (!webrtc->ap) {
+ pa_log_error("Failed to create audio processing");
+ goto fail;
+ }
+
+ webrtc->ap->echo_cancellation()->Enable(true);
+ webrtc->ap->noise_suppression()->Enable(true);
+ webrtc->ap->noise_suppression()->set_level(static_cast<NoiseSuppression::Level>(1));
+
+ webrtc->ap->gain_control()->set_mode(GainControl::kAdaptiveDigital);
+ //webrtc->ap->gain_control()->set_target_level_dbfs(30);
+ //webrtc->ap->gain_control()->set_stream_analog_level(30);
+ webrtc->ap->echo_cancellation()->set_suppression_level(static_cast<EchoCancellation::SuppressionLevel>(1));
+
+ webrtc->sconfig = new StreamConfig(ss->rate, ss->channels, false);
+ if (!webrtc->sconfig) {
+ pa_log_error("Failed to create stream config");
+ goto fail;
+ }
+
+ webrtc->sconfig->set_sample_rate_hz(ss->rate);
+ webrtc->sconfig->set_num_channels(ss->channels);
+
+ /* webrtc supports 10ms by default */
+ webrtc->frames = webrtc->sconfig->num_frames();
+ if (webrtc->frames != webrtc->fixed_frames) {
+ pa_log_error("Failed to set frames. frames(%zu), fixed_frames(%zu)",
+ webrtc->frames, webrtc->fixed_frames);
+ goto fail;
+ }
+
+ allocate_stream_buffer(webrtc, webrtc->fixed_frames);
+
+ pa_log_info("webrtc processes request block(%llumsec) block(%llumsec) n(%d)\n",
+ pa_bytes_to_usec(request_bytes, ss) / PA_USEC_PER_MSEC,
+ pa_bytes_to_usec(fixed_bytes, ss) / PA_USEC_PER_MSEC,
+ webrtc->loops);
+
+ return webrtc;
+
+fail:
+ if (webrtc->ap)
+ delete webrtc->ap;
+ if (webrtc->sconfig)
+ delete webrtc->sconfig;
+
+ pa_xfree(webrtc);
+
+ return NULL;
+}
+
+int32_t webrtc_audio_process(void *priv, int8_t *rec, int8_t *ref, int8_t *out) {
+ struct algo_webrtc *webrtc = (struct algo_webrtc *)priv;
+ pa_sample_spec ss;
+ size_t frames;
+
+ pa_assert(webrtc);
+ pa_assert(rec);
+ pa_assert(ref);
+ pa_assert(out);
+
+ ss.format = PA_SAMPLE_FLOAT32LE;
+ ss.rate = webrtc->ss.rate;
+ ss.channels = webrtc->ss.channels;
+
+ frames = webrtc->fixed_frames;
+
+ for (int i = 0; i < webrtc->loops; i++) {
+ int ret;
+
+ convert_s16_to_float(webrtc->ref_fbuf, (int16_t *)ref, frames * ss.channels);
+ pa_deinterleave(webrtc->ref_fbuf, (void **)webrtc->ref_dbuf, ss.channels,
+ pa_sample_size(&ss), frames);
+
+ /* reference */
+ ret = webrtc->ap->ProcessReverseStream(webrtc->ref_dbuf,
+ *webrtc->sconfig,
+ *webrtc->sconfig,
+ webrtc->ref_dbuf);
+ if (ret != AudioProcessing::kNoError) {
+ pa_log_error("Failed to process reverse stream");
+ return -1;
+ }
+
+ webrtc->ap->set_stream_delay_ms(0);
+
+ /* capture */
+ convert_s16_to_float(webrtc->rec_fbuf, (int16_t *)rec, frames * ss.channels);
+ pa_deinterleave(webrtc->rec_fbuf, (void **)webrtc->rec_dbuf, ss.channels,
+ pa_sample_size(&ss), frames);
+
+ ret = webrtc->ap->ProcessStream(webrtc->rec_dbuf,
+ *webrtc->sconfig,
+ *webrtc->sconfig,
+ webrtc->out_dbuf);
+ if (ret != AudioProcessing::kNoError) {
+ pa_log_error("Failed to process stream");
+ return -1;
+ }
+
+ pa_interleave((const void **)webrtc->out_dbuf, ss.channels, webrtc->out_fbuf,
+ pa_sample_size(&ss), frames);
+ convert_float_to_s16((int16_t *)out, webrtc->out_fbuf, frames * ss.channels);
+
+ rec += webrtc->fixed_bytes;
+ ref += webrtc->fixed_bytes;
+ out += webrtc->fixed_bytes;
+ }
+
+ return 0;
+}
+
+int32_t webrtc_audio_destroy(void *priv) {
+ struct algo_webrtc *webrtc = (struct algo_webrtc *)priv;
+
+ pa_assert(webrtc);
+
+ delete webrtc->sconfig;
+ delete webrtc->ap;
+
+ deallocate_stream_buffer(webrtc);
+ pa_xfree(webrtc);
+
+ return 0;
+}
+
+static void deallocate_stream_buffer(struct algo_webrtc *webrtc) {
+ pa_assert(webrtc);
+
+ for (int i = 0; i < webrtc->ss.channels; i++) {
+ pa_xfree(webrtc->rec_dbuf[i]);
+ pa_xfree(webrtc->ref_dbuf[i]);
+ pa_xfree(webrtc->out_dbuf[i]);
+ }
+
+ pa_xfree(webrtc->rec_fbuf);
+ pa_xfree(webrtc->ref_fbuf);
+ pa_xfree(webrtc->out_fbuf);
+}
+
+static void allocate_stream_buffer(struct algo_webrtc *webrtc, size_t nframes) {
+ int channels;
+
+ pa_assert(webrtc);
+
+ channels = webrtc->ss.channels;
+
+ webrtc->rec_fbuf = pa_xnew(float, nframes * channels);
+ webrtc->ref_fbuf = pa_xnew(float, nframes * channels);
+ webrtc->out_fbuf = pa_xnew(float, nframes * channels);
+
+ for (int i = 0; i < channels; i++) {
+ webrtc->rec_dbuf[i] = pa_xnew(float, nframes);
+ webrtc->ref_dbuf[i] = pa_xnew(float, nframes);
+ webrtc->out_dbuf[i] = pa_xnew(float, nframes);
+ }
+}
+
+static void convert_s16_to_float(float *dst, int16_t *src, size_t n) {
+ pa_assert(dst);
+ pa_assert(src);
+
+ ((pa_convert_func_t)pa_get_convert_to_float32ne_function(PA_SAMPLE_S16LE))(n, src, dst);
+}
+
+static void convert_float_to_s16(int16_t *dst, float *src, size_t n) {
+ pa_assert(dst);
+ pa_assert(src);
+
+ ((pa_convert_func_t)pa_get_convert_to_s16ne_function(PA_SAMPLE_FLOAT32LE))(n, src, dst);
+}
+
projects / platform / core / multimedia / pulseaudio-modules-tizen.git / commitdiff