pool/pulseaudio
SHA256
5
0
Fork 1
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Accepting request 1112006 from home:tiwai:branches:multimedia:libs

Browse Source 
- Enable webrtc AEC3 support:
  echo-cancel-add-webrtc-AEC3-support.patch
- Build fixes for webrtc-audio-processing 1.3 (only enabled for TW,
  so far; Leap 15.x still receives the old version):
  build-sys-Bump-cpp_std-to-c-17.patch
  build-sys-Bump-webrtc-audio-processing-dependency.patch

OBS-URL: https://build.opensuse.org/request/show/1112006
OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/pulseaudio?expand=0&rev=263
This commit is contained in:
Takashi Iwai 2023-09-18 14:57:51 +00:00 committed by Git OBS Bridge
parent 87eb029056
commit c2e0c759b4
5 changed files with 695 additions and 0 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

23
build-sys-Bump-cpp_std-to-c-17.patch Normal file
View File

@ -0,0 +1,23 @@
From e2b63d157fcc5ceb67a2f0eaed202d18baa05a11 Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Sun, 13 Aug 2023 07:24:41 -0400
Subject: [PATCH] build-sys: Bump cpp_std to c++17
Match it with webrtc-audio-processing, which is what we care about.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/795>
---
meson.build | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
--- a/meson.build
+++ b/meson.build
@@ -1,7 +1,7 @@
project('pulseaudio', 'c', 'cpp',
version : run_command(find_program('git-version-gen'), join_paths(meson.current_source_dir(), '.tarball-version')).stdout().strip(),
meson_version : '>= 0.50.0',
- default_options : [ 'c_std=gnu11', 'cpp_std=c++11' ]
+ default_options : [ 'c_std=gnu11', 'cpp_std=c++17' ]
)
meson.add_dist_script('scripts/save-tarball-version.sh', meson.project_version())

29
build-sys-Bump-webrtc-audio-processing-dependency.patch Normal file
View File

@ -0,0 +1,29 @@
From 84c53066c65439deb42d29bba8c6899a4fa0e318 Mon Sep 17 00:00:00 2001
From: Arun Raghavan <arun@asymptotic.io>
Date: Tue, 20 Oct 2020 17:29:55 -0400
Subject: [PATCH] build-sys: Bump webrtc-audio-processing dependency
The package name and versioning are changing upstream, so prepare for
that.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/395>
---
meson.build | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/meson.build b/meson.build
index b678bb531aad..a1652e4d3094 100644
--- a/meson.build
+++ b/meson.build
@@ -728,7 +728,7 @@ if get_option('daemon')
cdata.set('HAVE_SOXR', 1)
endif
- webrtc_dep = dependency('webrtc-audio-processing', version : '>= 0.2', required : get_option('webrtc-aec'))
+ webrtc_dep = dependency('webrtc-audio-processing-1', version : '>= 1.0', required : get_option('webrtc-aec'))
if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
endif
--
2.35.3

617
echo-cancel-add-webrtc-AEC3-support.patch Normal file
View File

@ -0,0 +1,617 @@
From 22bbb5b3ba0d28d630b10944fe19d7f9eee3a00f Mon Sep 17 00:00:00 2001
From: Eero Nurkkala <eero.nurkkala@offcode.fi>
Date: Tue, 20 Oct 2020 16:20:23 -0400
Subject: [PATCH] echo-cancel: add webrtc AEC3 support
Drop a number of now unsupported features, and add new parameters for
pre-/post-amplification.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/395>
---
src/modules/echo-cancel/webrtc.cc | 437 ++++++++----------------------
1 file changed, 115 insertions(+), 322 deletions(-)
diff --git a/src/modules/echo-cancel/webrtc.cc b/src/modules/echo-cancel/webrtc.cc
index 56daab0fd05e..ed4bb65a56a5 100644
--- a/src/modules/echo-cancel/webrtc.cc
+++ b/src/modules/echo-cancel/webrtc.cc
@@ -3,8 +3,8 @@
Copyright 2011 Collabora Ltd.
2015 Aldebaran SoftBank Group
-
- Contributor: Arun Raghavan <mail@arunraghavan.net>
+ 2020 Arun Raghavan <arun@asymptotic.io>
+ 2020 Eero Nurkkala <eero.nurkkala@offcode.fi>
PulseAudio is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as published
@@ -34,80 +34,47 @@ PA_C_DECL_BEGIN
#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 WEBRTC_APM_DEBUG_DUMP 0
+
+#include <modules/audio_processing/include/audio_processing.h>
#define BLOCK_SIZE_US 10000
#define DEFAULT_HIGH_PASS_FILTER true
#define DEFAULT_NOISE_SUPPRESSION true
+#define DEFAULT_TRANSIENT_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_VAD false
#define DEFAULT_AGC_START_VOLUME 85
-#define DEFAULT_BEAMFORMING false
-#define DEFAULT_TRACE false
+#define DEFAULT_POSTAMP_ENABLE false
+#define DEFAULT_POSTAMP_GAIN_DB 0
+#define DEFAULT_PREAMP_ENABLE false
+#define DEFAULT_PREAMP_GAIN_DB 0
#define WEBRTC_AGC_MAX_VOLUME 255
+#define WEBRTC_POSTAMP_GAIN_MAX_DB 90
+#define WEBRTC_PREAMP_GAIN_MAX_DB 90
static const char* const valid_modargs[] = {
- "high_pass_filter",
- "noise_suppression",
+ "agc_start_volume",
"analog_gain_control",
"digital_gain_control",
+ "high_pass_filter",
"mobile",
- "routing_mode",
- "comfort_noise",
- "drift_compensation",
+ "noise_suppression",
+ "post_amplifier",
+ "post_amplifier_gain",
+ "pre_amplifier",
+ "pre_amplifier_gain",
+ "transient_noise_suppression",
"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("%s", message);
- else if (level & webrtc::kTraceWarning)
- pa_log_warn("%s", message);
- else if (level & webrtc::kTraceInfo)
- pa_log_info("%s", message);
- else
- pa_log_debug("%s", message);
- }
-};
-
static int webrtc_volume_from_pa(pa_volume_t v)
{
return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM;
@@ -120,8 +87,7 @@ static pa_volume_t webrtc_volume_to_pa(int v)
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)
+ pa_sample_spec *out_ss, pa_channel_map *out_map)
{
rec_ss->format = PA_SAMPLE_FLOAT32NE;
play_ss->format = PA_SAMPLE_FLOAT32NE;
@@ -139,110 +105,22 @@ static void webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_ma
*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::AudioProcessing *apm = webrtc::AudioProcessingBuilder().Create();
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;
+ webrtc::AudioProcessing::Config config;
+ bool hpf, ns, tns, agc, dgc, mobile, pre_amp, vad, post_amp;
+ int i;
+ uint32_t agc_start_volume, pre_amp_gain, post_amp_gain;
pa_modargs *ma;
- bool trace = false;
if (!(ma = pa_modargs_new(args, valid_modargs))) {
pa_log("Failed to parse submodule arguments.");
@@ -261,6 +139,12 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
goto fail;
}
+ tns = DEFAULT_TRANSIENT_NOISE_SUPPRESSION;
+ if (pa_modargs_get_value_boolean(ma, "transient_noise_suppression", &tns) < 0) {
+ pa_log("Failed to parse transient_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");
@@ -278,6 +162,36 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
goto fail;
}
+ pre_amp = DEFAULT_PREAMP_ENABLE;
+ if (pa_modargs_get_value_boolean(ma, "pre_amplifier", &pre_amp) < 0) {
+ pa_log("Failed to parse pre_amplifier value");
+ goto fail;
+ }
+ pre_amp_gain = DEFAULT_PREAMP_GAIN_DB;
+ if (pa_modargs_get_value_u32(ma, "pre_amplifier_gain", &pre_amp_gain) < 0) {
+ pa_log("Failed to parse pre_amplifier_gain value");
+ goto fail;
+ }
+ if (pre_amp_gain > WEBRTC_PREAMP_GAIN_MAX_DB) {
+ pa_log("Preamp gain must not exceed %u", WEBRTC_PREAMP_GAIN_MAX_DB);
+ goto fail;
+ }
+
+ post_amp = DEFAULT_POSTAMP_ENABLE;
+ if (pa_modargs_get_value_boolean(ma, "post_amplifier", &post_amp) < 0) {
+ pa_log("Failed to parse post_amplifier value");
+ goto fail;
+ }
+ post_amp_gain = DEFAULT_POSTAMP_GAIN_DB;
+ if (pa_modargs_get_value_u32(ma, "post_amplifier_gain", &post_amp_gain) < 0) {
+ pa_log("Failed to parse post_amplifier_gain value");
+ goto fail;
+ }
+ if (post_amp_gain > WEBRTC_POSTAMP_GAIN_MAX_DB) {
+ pa_log("Postamp gain must not exceed %u", WEBRTC_POSTAMP_GAIN_MAX_DB);
+ goto fail;
+ }
+
mobile = DEFAULT_MOBILE;
if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
pa_log("Failed to parse mobile value");
@@ -285,33 +199,6 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
}
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) {
@@ -319,24 +206,6 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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");
@@ -348,82 +217,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
}
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);
+ webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map);
pconfig = {
webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */
@@ -436,46 +230,60 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
goto fail;
}
+ if (pre_amp) {
+ config.pre_amplifier.enabled = true;
+ config.pre_amplifier.fixed_gain_factor = (float)pre_amp_gain;
+ } else
+ config.pre_amplifier.enabled = false;
+
if (hpf)
- apm->high_pass_filter()->Enable(true);
+ config.high_pass_filter.enabled = true;
+ else
+ config.high_pass_filter.enabled = false;
- 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);
- }
+ config.echo_canceller.enabled = true;
- if (ns) {
- apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
- apm->noise_suppression()->Enable(true);
- }
+ if (!mobile)
+ config.echo_canceller.mobile_mode = false;
+ else
+ config.echo_canceller.mobile_mode = 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;
- }
+ if (ns)
+ config.noise_suppression.enabled = true;
+ else
+ config.noise_suppression.enabled = false;
- apm->gain_control()->Enable(true);
+ if (tns)
+ config.transient_suppression.enabled = true;
+ else
+ config.transient_suppression.enabled = false;
+
+ if (dgc) {
+ ec->params.webrtc.agc = false;
+ config.gain_controller1.enabled = true;
+ if (mobile)
+ config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kFixedDigital;
+ else
+ config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
+ } else if (agc) {
+ ec->params.webrtc.agc = true;
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ config.gain_controller1.analog_level_minimum = 0;
+ config.gain_controller1.analog_level_maximum = WEBRTC_AGC_MAX_VOLUME;
}
if (vad)
- apm->voice_detection()->Enable(true);
+ config.voice_detection.enabled = true;
+ else
+ config.voice_detection.enabled = false;
+
+ if (post_amp) {
+ config.gain_controller2.enabled = true;
+ config.gain_controller2.fixed_digital.gain_db = (float)post_amp_gain;
+ config.gain_controller2.adaptive_digital.enabled = false;
+ } else
+ config.gain_controller2.enabled = false;
ec->params.webrtc.apm = apm;
ec->params.webrtc.rec_ss = *rec_ss;
@@ -485,6 +293,8 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
*nframes = ec->params.webrtc.blocksize;
ec->params.webrtc.first = true;
+ apm->ApplyConfig(config);
+
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++)
@@ -496,10 +306,7 @@ bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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)
+ if (apm)
delete apm;
return false;
@@ -515,12 +322,6 @@ void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
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) {
@@ -538,7 +339,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
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_analog_level(old_volume);
}
apm->set_stream_delay_ms(0);
@@ -553,7 +354,7 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
ec->params.webrtc.first = false;
new_volume = ec->params.webrtc.agc_start_volume;
} else {
- new_volume = apm->gain_control()->stream_analog_level();
+ new_volume = apm->recommended_stream_analog_level();
}
if (old_volume != new_volume)
@@ -564,9 +365,6 @@ void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out
}
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) {
@@ -577,11 +375,6 @@ 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);
- }
-
if (ec->params.webrtc.apm) {
delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
ec->params.webrtc.apm = NULL;
--
2.35.3

10
pulseaudio.changes
View File

@ -1,3 +1,13 @@
-------------------------------------------------------------------
Mon Sep 18 14:33:40 UTC 2023 - Takashi Iwai <tiwai@suse.com>
- Enable webrtc AEC3 support:
echo-cancel-add-webrtc-AEC3-support.patch
- Build fixes for webrtc-audio-processing 1.3 (only enabled for TW,
so far; Leap 15.x still receives the old version):
build-sys-Bump-cpp_std-to-c-17.patch
build-sys-Bump-webrtc-audio-processing-dependency.patch
-------------------------------------------------------------------
Fri Jun 30 11:01:11 UTC 2023 - Fabian Vogt <fvogt@suse.com>

16
pulseaudio.spec
View File

@ -25,6 +25,11 @@
%define _fillupdir /var/adm/fillup-templates
%endif
# temporary workaround for missing webrtc-audio-processing 1.3 on Leap 15.x
%if 0%{?suse_version} < 1599
%define with_old_webrtc 1
%endif
%define drvver 16.1
%define soname 0
%define _udevrulesdir %(pkg-config --variable=udevdir udev)/rules.d
@ -54,6 +59,12 @@ Patch5: qpaeq-shebang.patch
Patch6: pulseaudio-old-systemd-workaround.patch
# PATCH-FIX-OPENSUSE Workaround for suse-module-tools directory
Patch7: pulseaudio-dump-module-Ignore-invalid-module-init-tools.patch
# PATCH-FIX-UPSTREAM fix for webrtc-audioprocessing 1.3
Patch8: echo-cancel-add-webrtc-AEC3-support.patch
# PATCH-FIX-UPSTREAM fix for webrtc-audioprocessing 1.3
Patch9: build-sys-Bump-cpp_std-to-c-17.patch
# PATCH-FIX-UPSTREAM fix for webrtc-audioprocessing 1.3
Patch10: build-sys-Bump-webrtc-audio-processing-dependency.patch
BuildRequires: alsa-devel >= 1.0.19
BuildRequires: bluez-devel >= 5
BuildRequires: fdupes
@ -330,6 +341,11 @@ System user for PulseAudio
%patch6 -p1
%endif
%patch7 -p1
%if !0%{?with_old_webrtc}
%patch8 -p1
%patch9 -p1
%patch10 -p1
%endif
%build
%meson \