diff options
Diffstat (limited to 'src/modules/echo-cancel/webrtc.cc')
| -rw-r--r-- | src/modules/echo-cancel/webrtc.cc | 594 |
1 files changed, 594 insertions, 0 deletions
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc new file mode 100644 index 0000000..ec3ba06 --- /dev/null +++ b/src/modules/echo-cancel/webrtc.cc @@ -0,0 +1,594 @@ +/*** + This file is part of PulseAudio. + + Copyright 2011 Collabora Ltd. + 2015 Aldebaran SoftBank Group + + Contributor: Arun Raghavan <mail@arunraghavan.net> + + 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, see <http://www.gnu.org/licenses/>. +***/ + +#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 <webrtc/modules/audio_processing/include/audio_processing.h> +#include <webrtc/modules/interface/module_common_types.h> +#include <webrtc/system_wrappers/include/trace.h> + +#define BLOCK_SIZE_US 10000 + +#define DEFAULT_HIGH_PASS_FILTER true +#define DEFAULT_NOISE_SUPPRESSION true +#define DEFAULT_ANALOG_GAIN_CONTROL true +#define DEFAULT_DIGITAL_GAIN_CONTROL false +#define DEFAULT_MOBILE false +#define DEFAULT_ROUTING_MODE "speakerphone" +#define DEFAULT_COMFORT_NOISE true +#define DEFAULT_DRIFT_COMPENSATION false +#define DEFAULT_VAD true +#define DEFAULT_EXTENDED_FILTER false +#define DEFAULT_INTELLIGIBILITY_ENHANCER false +#define DEFAULT_EXPERIMENTAL_AGC false +#define DEFAULT_AGC_START_VOLUME 85 +#define DEFAULT_BEAMFORMING false +#define DEFAULT_TRACE false + +#define WEBRTC_AGC_MAX_VOLUME 255 + +static const char* const valid_modargs[] = { + "high_pass_filter", + "noise_suppression", + "analog_gain_control", + "digital_gain_control", + "mobile", + "routing_mode", + "comfort_noise", + "drift_compensation", + "voice_detection", + "extended_filter", + "intelligibility_enhancer", + "experimental_agc", + "agc_start_volume", + "beamforming", + "mic_geometry", /* documented in parse_mic_geometry() */ + "target_direction", /* documented in parse_mic_geometry() */ + "trace", + 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; +} + +class PaWebrtcTraceCallback : public webrtc::TraceCallback { + void Print(webrtc::TraceLevel level, const char *message, int length) + { + if (level & webrtc::kTraceError || level & webrtc::kTraceCritical) + pa_log(message); + else if (level & webrtc::kTraceWarning) + pa_log_warn(message); + else if (level & webrtc::kTraceInfo) + pa_log_info(message); + else + pa_log_debug(message); + } +}; + +static int webrtc_volume_from_pa(pa_volume_t v) +{ + return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM; +} + +static pa_volume_t webrtc_volume_to_pa(int v) +{ + return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME; +} + +static void webrtc_ec_fixate_spec(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, + bool beamforming) +{ + rec_ss->format = PA_SAMPLE_FLOAT32NE; + play_ss->format = PA_SAMPLE_FLOAT32NE; + + /* AudioProcessing expects one of the following rates */ + if (rec_ss->rate >= 48000) + rec_ss->rate = 48000; + else if (rec_ss->rate >= 32000) + rec_ss->rate = 32000; + else if (rec_ss->rate >= 16000) + rec_ss->rate = 16000; + else + rec_ss->rate = 8000; + + *out_ss = *rec_ss; + *out_map = *rec_map; + + if (beamforming) { + /* The beamformer gives us a single channel */ + out_ss->channels = 1; + pa_channel_map_init_mono(out_map); + } + + /* Playback stream rate needs to be the same as capture */ + play_ss->rate = rec_ss->rate; +} + +static bool parse_point(const char **point, float (&f)[3]) { + int ret, length; + + ret = sscanf(*point, "%g,%g,%g%n", &f[0], &f[1], &f[2], &length); + if (ret != 3) + return false; + + /* Consume the bytes we've read so far */ + *point += length; + + return true; +} + +static bool parse_mic_geometry(const char **mic_geometry, std::vector<webrtc::Point>& geometry) { + /* The microphone geometry is expressed as cartesian point form: + * x1,y1,z1,x2,y2,z2,... + * + * Where x1,y1,z1 is the position of the first microphone with regards to + * the array's "center", x2,y2,z2 the position of the second, and so on. + * + * 'x' is the horizontal coordinate, with positive values being to the + * right from the mic array's perspective. + * + * 'y' is the depth coordinate, with positive values being in front of the + * array. + * + * 'z' is the vertical coordinate, with positive values being above the + * array. + * + * All distances are in meters. + */ + + /* The target direction is expected to be in spherical point form: + * a,e,r + * + * Where 'a' is the azimuth of the target point relative to the center of + * the array, 'e' its elevation, and 'r' the radius. + * + * 0 radians azimuth is to the right of the array, and positive angles + * move in a counter-clockwise direction. + * + * 0 radians elevation is horizontal w.r.t. the array, and positive + * angles go upwards. + * + * radius is distance from the array center in meters. + */ + + long unsigned int i; + float f[3]; + + for (i = 0; i < geometry.size(); i++) { + if (!parse_point(mic_geometry, f)) { + pa_log("Failed to parse channel %lu in mic_geometry", i); + return false; + } + + /* Except for the last point, we should have a trailing comma */ + if (i != geometry.size() - 1) { + if (**mic_geometry != ',') { + pa_log("Failed to parse channel %lu in mic_geometry", i); + return false; + } + + (*mic_geometry)++; + } + + pa_log_debug("Got mic #%lu position: (%g, %g, %g)", i, f[0], f[1], f[2]); + + geometry[i].c[0] = f[0]; + geometry[i].c[1] = f[1]; + geometry[i].c[2] = f[2]; + } + + if (**mic_geometry != '\0') { + pa_log("Failed to parse mic_geometry value: more parameters than expected"); + return false; + } + + return true; +} + +bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec, + 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; + webrtc::ProcessingConfig pconfig; + webrtc::Config config; + bool hpf, ns, agc, dgc, mobile, cn, vad, ext_filter, intelligibility, experimental_agc, beamforming; + int rm = -1, i; + uint32_t agc_start_volume; + pa_modargs *ma; + bool trace = false; + + 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 = agc ? false : DEFAULT_DIGITAL_GAIN_CONTROL; + if (pa_modargs_get_value_boolean(ma, "digital_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; + } + + ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION; + if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) { + pa_log("Failed to parse drift_compensation value"); + goto fail; + } + + if (mobile) { + if (ec->params.drift_compensation) { + pa_log("Can't use drift_compensation in mobile mode"); + goto fail; + } + + 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; + } + } + + vad = DEFAULT_VAD; + if (pa_modargs_get_value_boolean(ma, "voice_detection", &vad) < 0) { + pa_log("Failed to parse voice_detection value"); + goto fail; + } + + ext_filter = DEFAULT_EXTENDED_FILTER; + if (pa_modargs_get_value_boolean(ma, "extended_filter", &ext_filter) < 0) { + pa_log("Failed to parse extended_filter value"); + goto fail; + } + + intelligibility = DEFAULT_INTELLIGIBILITY_ENHANCER; + if (pa_modargs_get_value_boolean(ma, "intelligibility_enhancer", &intelligibility) < 0) { + pa_log("Failed to parse intelligibility_enhancer value"); + goto fail; + } + + experimental_agc = DEFAULT_EXPERIMENTAL_AGC; + if (pa_modargs_get_value_boolean(ma, "experimental_agc", &experimental_agc) < 0) { + pa_log("Failed to parse experimental_agc value"); + goto fail; + } + + agc_start_volume = DEFAULT_AGC_START_VOLUME; + if (pa_modargs_get_value_u32(ma, "agc_start_volume", &agc_start_volume) < 0) { + pa_log("Failed to parse agc_start_volume value"); + goto fail; + } + if (agc_start_volume > WEBRTC_AGC_MAX_VOLUME) { + pa_log("AGC start volume must not exceed %u", WEBRTC_AGC_MAX_VOLUME); + goto fail; + } + ec->params.webrtc.agc_start_volume = agc_start_volume; + + beamforming = DEFAULT_BEAMFORMING; + if (pa_modargs_get_value_boolean(ma, "beamforming", &beamforming) < 0) { + pa_log("Failed to parse beamforming value"); + goto fail; + } + + if (ext_filter) + config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true)); + if (intelligibility) + pa_log_warn("The intelligibility enhancer is not currently supported"); + if (experimental_agc) + config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, ec->params.webrtc.agc_start_volume)); + + trace = DEFAULT_TRACE; + if (pa_modargs_get_value_boolean(ma, "trace", &trace) < 0) { + pa_log("Failed to parse trace value"); + goto fail; + } + + if (trace) { + webrtc::Trace::CreateTrace(); + webrtc::Trace::set_level_filter(webrtc::kTraceAll); + ec->params.webrtc.trace_callback = new PaWebrtcTraceCallback(); + webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback); + } + + webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map, beamforming); + + /* We do this after fixate because we need the capture channel count */ + if (beamforming) { + std::vector<webrtc::Point> geometry(rec_ss->channels); + webrtc::SphericalPointf direction(0.0f, 0.0f, 0.0f); + const char *mic_geometry, *target_direction; + + if (!(mic_geometry = pa_modargs_get_value(ma, "mic_geometry", NULL))) { + pa_log("mic_geometry must be set if beamforming is enabled"); + goto fail; + } + + if (!parse_mic_geometry(&mic_geometry, geometry)) { + pa_log("Failed to parse mic_geometry value"); + goto fail; + } + + if ((target_direction = pa_modargs_get_value(ma, "target_direction", NULL))) { + float f[3]; + + if (!parse_point(&target_direction, f)) { + pa_log("Failed to parse target_direction value"); + goto fail; + } + + if (*target_direction != '\0') { + pa_log("Failed to parse target_direction value: more parameters than expected"); + goto fail; + } + +#define IS_ZERO(f) ((f) < 0.000001 && (f) > -0.000001) + + if (!IS_ZERO(f[1]) || !IS_ZERO(f[2])) { + pa_log("The beamformer currently only supports targeting along the azimuth"); + goto fail; + } + + direction.s[0] = f[0]; + direction.s[1] = f[1]; + direction.s[2] = f[2]; + } + + if (!target_direction) + config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry)); + else + config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry, direction)); + } + + apm = webrtc::AudioProcessing::Create(config); + + pconfig = { + webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */ + webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* output stream */ + webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */ + webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */ + }; + if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) { + pa_log("Error initialising audio processing module"); + goto fail; + } + + if (hpf) + apm->high_pass_filter()->Enable(true); + + if (!mobile) { + apm->echo_cancellation()->enable_drift_compensation(ec->params.drift_compensation); + 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); + ec->params.webrtc.agc = false; + } else if (dgc) { + apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital); + ec->params.webrtc.agc = false; + } else { + apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog); + if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) != + webrtc::AudioProcessing::kNoError) { + pa_log("Failed to initialise AGC"); + goto fail; + } + ec->params.webrtc.agc = true; + } + + apm->gain_control()->Enable(true); + } + + if (vad) + apm->voice_detection()->Enable(true); + + ec->params.webrtc.apm = apm; + ec->params.webrtc.rec_ss = *rec_ss; + ec->params.webrtc.play_ss = *play_ss; + ec->params.webrtc.out_ss = *out_ss; + ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC; + *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; + +fail: + if (ma) + pa_modargs_free(ma); + if (ec->params.webrtc.trace_callback) { + webrtc::Trace::ReturnTrace(); + delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback); + } if (apm) + delete apm; + + return false; +} + +void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) { + webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; + 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); + + pa_deinterleave(play, (void **) buf, ss->channels, pa_sample_size(ss), n); + + 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 + * effect. This function is called at the source side, but the processing + * would have to be done in the sink to be able to feed the processed audio + * to speakers. */ +} + +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; + 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; + 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); + + pa_deinterleave(rec, (void **) buf, rec_ss->channels, pa_sample_size(rec_ss), n); + + if (ec->params.webrtc.agc) { + pa_volume_t v = pa_echo_canceller_get_capture_volume(ec); + old_volume = webrtc_volume_from_pa(v); + apm->gain_control()->set_stream_analog_level(old_volume); + } + + apm->set_stream_delay_ms(0); + 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)) { + /* We start at a sane default volume (taken from the Chromium + * condition on the experimental AGC in audio_processing.h). This is + * needed to make sure that there's enough energy in the capture + * signal for the AGC to work */ + ec->params.webrtc.first = false; + new_volume = ec->params.webrtc.agc_start_volume; + } else { + new_volume = apm->gain_control()->stream_analog_level(); + } + + if (old_volume != new_volume) + pa_echo_canceller_set_capture_volume(ec, webrtc_volume_to_pa(new_volume)); + } + + 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) { + webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm; + + apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize); +} + +void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) { + pa_webrtc_ec_play(ec, play); + pa_webrtc_ec_record(ec, rec, out); +} + +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); + } + + if (ec->params.webrtc.apm) { + 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]); +} |