AC_PROG_CC
AC_PROG_CC_C99
AM_PROG_CC_C_O
+# Only required if you want the WebRTC canceller -- no runtime dep on
+# libstdc++ otherwise
+AC_PROG_CXX
AC_PROG_GCC_TRADITIONAL
AC_USE_SYSTEM_EXTENSIONS
fi
fi
+AC_ARG_ENABLE([webrtc-aec],
+ AS_HELP_STRING([--enable-webrtc-aec], [Enable the optional WebRTC-based echo canceller]))
+
+AS_IF([test "x$enable_webrtc_aec" != "xno"],
+ [PKG_CHECK_MODULES(WEBRTC, [ webrtc-audio-processing ], [HAVE_WEBRTC=1], [HAVE_WEBRTC=0])],
+ [HAVE_WEBRTC=0])
+
+AS_IF([test "x$enable_webrtc_aec" = "xyes" && test "x$HAVE_WEBRTC" = "x0"],
+ [AC_MSG_ERROR([*** webrtc-audio-processing library not found])])
+
+AC_SUBST(WEBRTC_CFLAGS)
+AC_SUBST(WEBRTC_LIBS)
+AM_CONDITIONAL([HAVE_WEBRTC], [test "x$HAVE_WEBRTC" = "x1"])
+
###################################
# Output #
AS_IF([test "x$HAVE_OPENSSL" = "x1"], ENABLE_OPENSSL=yes, ENABLE_OPENSSL=no)
AS_IF([test "x$HAVE_FFTW" = "x1"], ENABLE_FFTW=yes, ENABLE_FFTW=no)
AS_IF([test "x$HAVE_ORC" = "xyes"], ENABLE_ORC=yes, ENABLE_ORC=no)
+AS_IF([test "x$HAVE_WEBRTC" = "x1"], ENABLE_WEBRTC=yes, ENABLE_WEBRTC=no)
AS_IF([test "x$HAVE_TDB" = "x1"], ENABLE_TDB=yes, ENABLE_TDB=no)
AS_IF([test "x$HAVE_GDBM" = "x1"], ENABLE_GDBM=yes, ENABLE_GDBM=no)
AS_IF([test "x$HAVE_SIMPLEDB" = "x1"], ENABLE_SIMPLEDB=yes, ENABLE_SIMPLEDB=no)
Enable OpenSSL (for Airtunes): ${ENABLE_OPENSSL}
Enable fftw: ${ENABLE_FFTW}
Enable orc: ${ENABLE_ORC}
+ Enable WebRTC echo canceller: ${ENABLE_WEBRTC}
Database
tdb: ${ENABLE_TDB}
gdbm: ${ENABLE_GDBM}
--- /dev/null
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2011 Collabora Ltd.
+
+ Contributor: Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ 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>
+
+PA_C_DECL_BEGIN
+#include <pulsecore/core-util.h>
+#include <pulsecore/modargs.h>
+
+#include <pulse/timeval.h>
+#include "echo-cancel.h"
+PA_C_DECL_END
+
+#include <audio_processing.h>
+#include <module_common_types.h>
+
+#define BLOCK_SIZE_US 10000
+
+#define DEFAULT_HIGH_PASS_FILTER TRUE
+#define DEFAULT_NOISE_SUPPRESSION TRUE
+#define DEFAULT_ANALOG_GAIN_CONTROL FALSE
+#define DEFAULT_DIGITAL_GAIN_CONTROL TRUE
+#define DEFAULT_MOBILE FALSE
+#define DEFAULT_ROUTING_MODE "speakerphone"
+#define DEFAULT_COMFORT_NOISE TRUE
+
+static const char* const valid_modargs[] = {
+ "high_pass_filter",
+ "noise_suppression",
+ "analog_gain_control",
+ "digital_gain_control",
+ "mobile",
+ "routing_mode",
+ "comfort_noise",
+ NULL
+};
+
+static int routing_mode_from_string(const char *rmode) {
+ if (pa_streq(rmode, "quiet-earpiece-or-headset"))
+ return webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
+ else if (pa_streq(rmode, "earpiece"))
+ return webrtc::EchoControlMobile::kEarpiece;
+ else if (pa_streq(rmode, "loud-earpiece"))
+ return webrtc::EchoControlMobile::kLoudEarpiece;
+ else if (pa_streq(rmode, "speakerphone"))
+ return webrtc::EchoControlMobile::kSpeakerphone;
+ else if (pa_streq(rmode, "loud-speakerphone"))
+ return webrtc::EchoControlMobile::kLoudSpeakerphone;
+ else
+ return -1;
+}
+
+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,
+ uint32_t *blocksize, const char *args)
+{
+ webrtc::AudioProcessing *apm = NULL;
+ pa_bool_t hpf, ns, agc, dgc, mobile, cn;
+ int rm;
+ pa_modargs *ma;
+
+ if (!(ma = pa_modargs_new(args, valid_modargs))) {
+ pa_log("Failed to parse submodule arguments.");
+ goto fail;
+ }
+
+
+ hpf = DEFAULT_HIGH_PASS_FILTER;
+ if (pa_modargs_get_value_boolean(ma, "high_pass_filter", &hpf) < 0) {
+ pa_log("Failed to parse high_pass_filter value");
+ goto fail;
+ }
+
+ ns = DEFAULT_NOISE_SUPPRESSION;
+ if (pa_modargs_get_value_boolean(ma, "noise_suppression", &ns) < 0) {
+ pa_log("Failed to parse noise_suppression value");
+ goto fail;
+ }
+
+ agc = DEFAULT_ANALOG_GAIN_CONTROL;
+ if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &agc) < 0) {
+ pa_log("Failed to parse analog_gain_control value");
+ goto fail;
+ }
+
+ dgc = DEFAULT_DIGITAL_GAIN_CONTROL;
+ if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &dgc) < 0) {
+ pa_log("Failed to parse digital_gain_control value");
+ goto fail;
+ }
+
+ if (agc && dgc) {
+ pa_log("You must pick only one between analog and digital gain control");
+ goto fail;
+ }
+
+ mobile = DEFAULT_MOBILE;
+ if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
+ pa_log("Failed to parse mobile value");
+ goto fail;
+ }
+
+ if (mobile) {
+ if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
+ pa_log("Failed to parse routing_mode value");
+ goto fail;
+ }
+
+ cn = DEFAULT_COMFORT_NOISE;
+ if (pa_modargs_get_value_boolean(ma, "comfort_noise", &cn) < 0) {
+ pa_log("Failed to parse cn value");
+ goto fail;
+ }
+ } else {
+ if (pa_modargs_get_value(ma, "comfort_noise", NULL) || pa_modargs_get_value(ma, "routing_mode", NULL)) {
+ pa_log("The routing_mode and comfort_noise options are only valid with mobile=true");
+ goto fail;
+ }
+ }
+
+ apm = webrtc::AudioProcessing::Create(0);
+
+ source_ss->format = PA_SAMPLE_S16NE;
+ *sink_ss = *source_ss;
+ /* FIXME: the implementation actually allows a different number of
+ * source/sink channels. Do we want to support that? */
+ *sink_map = *source_map;
+
+ apm->set_sample_rate_hz(source_ss->rate);
+
+ apm->set_num_channels(source_ss->channels, source_ss->channels);
+ apm->set_num_reverse_channels(sink_ss->channels);
+
+ if (hpf)
+ apm->high_pass_filter()->Enable(true);
+
+ if (!mobile) {
+ apm->echo_cancellation()->enable_drift_compensation(false);
+ apm->echo_cancellation()->Enable(true);
+ } else {
+ apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
+ apm->echo_control_mobile()->enable_comfort_noise(cn);
+ apm->echo_control_mobile()->Enable(true);
+ }
+
+ if (ns) {
+ apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
+ apm->noise_suppression()->Enable(true);
+ }
+
+ if (agc || dgc) {
+ if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece)
+ /* Maybe this should be a knob, but we've got a lot of knobs already */
+ apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
+ else if (dgc)
+ apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
+ else {
+ /* FIXME: Hook up for analog AGC */
+ pa_log("Analog gain control isn't implemented yet -- using ditital gain control.");
+ apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
+ }
+ }
+
+ apm->voice_detection()->Enable(true);
+
+ ec->params.priv.webrtc.apm = apm;
+ ec->params.priv.webrtc.sample_spec = *source_ss;
+ ec->params.priv.webrtc.blocksize = *blocksize = (uint64_t)pa_bytes_per_second(source_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
+
+ pa_modargs_free(ma);
+ return TRUE;
+
+fail:
+ if (ma)
+ pa_modargs_free(ma);
+ if (apm)
+ webrtc::AudioProcessing::Destroy(apm);
+
+ return FALSE;
+}
+
+void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
+ webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
+ webrtc::AudioFrame play_frame, out_frame;
+ const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
+
+ play_frame._audioChannel = ss->channels;
+ play_frame._frequencyInHz = ss->rate;
+ play_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
+ memcpy(play_frame._payloadData, play, ec->params.priv.webrtc.blocksize);
+
+ out_frame._audioChannel = ss->channels;
+ out_frame._frequencyInHz = ss->rate;
+ out_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
+ memcpy(out_frame._payloadData, rec, ec->params.priv.webrtc.blocksize);
+
+ apm->AnalyzeReverseStream(&play_frame);
+ apm->set_stream_delay_ms(0);
+ apm->ProcessStream(&out_frame);
+
+ memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
+}
+
+void pa_webrtc_ec_done(pa_echo_canceller *ec) {
+ if (ec->params.priv.webrtc.apm) {
+ webrtc::AudioProcessing::Destroy((webrtc::AudioProcessing*)ec->params.priv.webrtc.apm);
+ ec->params.priv.webrtc.apm = NULL;
+ }
+}