diff --git a/build-sys-Bump-cpp_std-to-c-17.patch b/build-sys-Bump-cpp_std-to-c-17.patch deleted file mode 100644 index 5baf2a4..0000000 --- a/build-sys-Bump-cpp_std-to-c-17.patch +++ /dev/null @@ -1,23 +0,0 @@ -From e2b63d157fcc5ceb67a2f0eaed202d18baa05a11 Mon Sep 17 00:00:00 2001 -From: Arun Raghavan -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: ---- - 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()) diff --git a/build-sys-Bump-webrtc-audio-processing-dependency.patch b/build-sys-Bump-webrtc-audio-processing-dependency.patch deleted file mode 100644 index b8a85fd..0000000 --- a/build-sys-Bump-webrtc-audio-processing-dependency.patch +++ /dev/null @@ -1,29 +0,0 @@ -From 84c53066c65439deb42d29bba8c6899a4fa0e318 Mon Sep 17 00:00:00 2001 -From: Arun Raghavan -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: ---- - 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 - diff --git a/echo-cancel-add-webrtc-AEC3-support.patch b/echo-cancel-add-webrtc-AEC3-support.patch deleted file mode 100644 index bfaae2f..0000000 --- a/echo-cancel-add-webrtc-AEC3-support.patch +++ /dev/null @@ -1,617 +0,0 @@ -From 22bbb5b3ba0d28d630b10944fe19d7f9eee3a00f Mon Sep 17 00:00:00 2001 -From: Eero Nurkkala -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: ---- - 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 -+ 2020 Arun Raghavan -+ 2020 Eero Nurkkala - - 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 --#include --#include -+#define WEBRTC_APM_DEBUG_DUMP 0 -+ -+#include - - #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& 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(new webrtc::ExtendedFilter(true)); -- if (intelligibility) -- pa_log_warn("The intelligibility enhancer is not currently supported"); -- if (experimental_agc) -- config.Set(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 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(new webrtc::Beamforming(true, geometry)); -- else -- config.Set(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(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 - diff --git a/pulseaudio-16.1.tar.xz b/pulseaudio-16.1.tar.xz deleted file mode 100644 index 7a63ae1..0000000 --- a/pulseaudio-16.1.tar.xz +++ /dev/null @@ -1,3 +0,0 @@ -version https://git-lfs.github.com/spec/v1 -oid sha256:8eef32ce91d47979f95fd9a935e738cd7eb7463430dabc72863251751e504ae4 -size 1545596 diff --git a/pulseaudio-17.0.tar.xz b/pulseaudio-17.0.tar.xz new file mode 100644 index 0000000..455a81d --- /dev/null +++ b/pulseaudio-17.0.tar.xz @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:053794d6671a3e397d849e478a80b82a63cb9d8ca296bd35b73317bb5ceb87b5 +size 1566556 diff --git a/pulseaudio.changes b/pulseaudio.changes index 46ff9aa..908ab61 100644 --- a/pulseaudio.changes +++ b/pulseaudio.changes @@ -1,3 +1,22 @@ +------------------------------------------------------------------- +Thu Feb 8 11:24:48 UTC 2024 - Takashi Iwai + +- Update to version 17.0: + * Updates to ALSA UCM-based setups + * Battery level indication to Bluetooth devices + * Support for the Bluetooth FastStream codec + * webrtc-audio-processing dependency updated + * Trigger role groups added to module-role-cork + * XDG base directory spec for profile-set loading + * PA_RATE_MAX increased + * webrtc-audio-processing dependency updated + For details, see: + https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/17.0/ +- Drop obsoleted patches: + echo-cancel-add-webrtc-AEC3-support.patch + build-sys-Bump-cpp_std-to-c-17.patch + build-sys-Bump-webrtc-audio-processing-dependency.patch + ------------------------------------------------------------------- Wed Dec 20 15:57:44 UTC 2023 - Giacomo Comes diff --git a/pulseaudio.spec b/pulseaudio.spec index 315e656..150dbac 100644 --- a/pulseaudio.spec +++ b/pulseaudio.spec @@ -25,12 +25,12 @@ %define _fillupdir /var/adm/fillup-templates %endif -%define drvver 16.1 +%define drvver 17.0 %define soname 0 %define _udevrulesdir %(pkg-config --variable=udevdir udev)/rules.d %define _bashcompletionsdir %{_datadir}/bash-completion/completions Name: pulseaudio -Version: 16.1 +Version: 17.0 Release: 0 Summary: A Networked Sound Server License: GPL-2.0-or-later AND LGPL-2.1-or-later @@ -54,12 +54,6 @@ 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 @@ -328,17 +322,14 @@ System user for PulseAudio %prep %setup -q -T -b0 -%patch0 -p1 -%patch1 -p1 -%patch5 -p1 +%patch -P0 -p1 +%patch -P1 -p1 +%patch -P5 -p1 # workaround for Leap 15.x %if 0%{?suse_version} < 1550 -%patch6 -p1 +%patch -P6 -p1 %endif -%patch7 -p1 -%patch8 -p1 -%patch9 -p1 -%patch10 -p1 +%patch -P7 -p1 %build %meson \ @@ -402,10 +393,6 @@ mkdir -p %{buildroot}%{_sysconfdir}/pulse/system.pa.d install -m 0644 %{SOURCE6} %{buildroot}%{_sysconfdir}/pulse/daemon.conf.d/60-disable_flat_volumes.conf # user install -Dm0644 %{SOURCE10} %{buildroot}%{_sysusersdir}/system-user-pulse.conf -# move dbus-1 system.d file to /usr -install -Dm0644 %{buildroot}%{_sysconfdir}/dbus-1/system.d/pulseaudio-system.conf %{buildroot}%{_datadir}/dbus-1/system.d/pulseaudio-system.conf -rm -rf %{buildroot}%{_sysconfdir}/dbus-1 - %find_lang %{name}