Merge branch 'master' of git://0pointer.de/pulseaudio

9 files changed, 259 insertions, 28 deletions

diff --git a/src/modules/alsa/mixer/paths/analog-input.conf.common b/src/modules/alsa/mixer/paths/analog-input.conf.common
index 87af38b3..951e11fa 100644
--- a/src/modules/alsa/mixer/paths/analog-input.conf.common
+++ b/src/modules/alsa/mixer/paths/analog-input.conf.common
@@ -98,7 +98,11 @@ priority = 18
 name = input-docking
 priority = 17
 
-;;; ' Capture Source'
+[Option Input Source:AUX IN]
+name = input
+priority = 10
+
+;;; 'Capture Source'
 
 [Element Capture Source]
 enumeration = select
@@ -244,6 +248,31 @@ name = input-docking
 [Option Capture Source:Dock Mic]
 name = input-docking-microphone
 
+;;; 'Mic Jack Mode'
+
+[Element Mic Jack Mode]
+enumeration = select
+
+[Option Mic Jack Mode:Mic In]
+name = input-microphone
+
+[Option Mic Jack Mode:Line In]
+name = input-linein
+
+;;; 'Digital Input Source'
+
+[Element Digital Input Source]
+enumeration = select
+
+[Option Digital Input Source:Analog Inputs]
+name = input
+
+[Option Digital Input Source:Digital Mic 1]
+name = input-microphone
+
+[Option Digital Input Source:Digital Mic 2]
+name = input-microphone
+
 ;;; Various Boosts
 
 [Element Capture Boost]
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
new file mode 100644
index 00000000..f2fd31c7
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
@@ -0,0 +1,82 @@
+# This file is part of PulseAudio.
+#
+# 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.
+
+; Path for mixers that have a 'Headphone2' control
+;
+; See analog-output.conf.common for an explanation on the directives
+
+[General]
+priority = 89
+
+[Element Hardware Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master Mono]
+switch = off
+volume = off
+
+; This profile path is intended to control the second headphones, not
+; the first headphones. But it should not hurt if we leave the
+; headphone jack enabled nonetheless.
+[Element Headphone]
+switch = mute
+volume = zero
+
+[Element Headphone2]
+required = any
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Speaker]
+switch = off
+volume = off
+
+[Element Front]
+switch = off
+volume = off
+
+[Element Rear]
+switch = off
+volume = off
+
+[Element Surround]
+switch = off
+volume = off
+
+[Element Side]
+switch = off
+volume = off
+
+[Element Center]
+switch = off
+volume = off
+
+[Element LFE]
+switch = off
+volume = off
+
+.include analog-output.conf.common
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
index 61d2e297..2131cfe8 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
@@ -44,6 +44,13 @@ volume = merge
 override-map.1 = all
 override-map.2 = all-left,all-right
 
+; This profile path is intended to control the first headphones, not
+; the second headphones. But it should not hurt if we leave the second
+; headphone jack enabled nonetheless.
+[Element Headphone2]
+switch = mute
+volume = zero
+
 [Element Speaker]
 switch = off
 volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
index 911361d7..0a43e271 100644
--- a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
@@ -48,6 +48,10 @@ override-map.2 = lfe,lfe
 switch = mute
 volume = zero
 
+[Element Headphone2]
+switch = mute
+volume = zero
+
 [Element Speaker]
 switch = mute
 volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-output-mono.conf b/src/modules/alsa/mixer/paths/analog-output-mono.conf
index 2fbc60b7..542edc40 100644
--- a/src/modules/alsa/mixer/paths/analog-output-mono.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-mono.conf
@@ -45,6 +45,10 @@ override-map.2 = all-left,all-right
 switch = mute
 volume = zero
 
+[Element Headphone2]
+switch = mute
+volume = zero
+
 [Element Speaker]
 switch = mute
 volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
new file mode 100644
index 00000000..aea78534
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
@@ -0,0 +1,94 @@
+# This file is part of PulseAudio.
+#
+# 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.
+
+; Path for mixers that have a 'Speaker' control
+;
+; See analog-output.conf.common for an explanation on the directives
+
+[General]
+priority = 100
+
+[Element Hardware Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master Mono]
+switch = off
+volume = off
+
+; This profile path is intended to control the speaker, not the
+; headphones. But it should not hurt if we leave the headphone jack
+; enabled nonetheless.
+[Element Headphone]
+switch = mute
+volume = zero
+
+[Element Headphone2]
+switch = mute
+volume = zero
+
+[Element Speaker]
+required = any
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Front]
+switch = mute
+volume = merge
+override-map.1 = all-front
+override-map.2 = front-left,front-right
+
+[Element Rear]
+switch = mute
+volume = merge
+override-map.1 = all-rear
+override-map.2 = rear-left,rear-right
+
+[Element Surround]
+switch = mute
+volume = merge
+override-map.1 = all-rear
+override-map.2 = rear-left,rear-right
+
+[Element Side]
+switch = mute
+volume = merge
+override-map.1 = all-side
+override-map.2 = side-left,side-right
+
+[Element Center]
+switch = mute
+volume = merge
+override-map.1 = all-center
+override-map.2 = all-center,all-center
+
+[Element LFE]
+switch = mute
+volume = merge
+override-map.1 = lfe
+override-map.2 = lfe,lfe
+
+.include analog-output.conf.common
diff --git a/src/modules/alsa/mixer/paths/analog-output.conf b/src/modules/alsa/mixer/paths/analog-output.conf
index f71a05a1..d7c1223b 100644
--- a/src/modules/alsa/mixer/paths/analog-output.conf
+++ b/src/modules/alsa/mixer/paths/analog-output.conf
@@ -14,12 +14,13 @@
 # along with PulseAudio; if not, write to the Free Software Foundation,
 # Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
 
-; Intended for the 'default' output
+; Intended for the 'default' output. Note that a-o-speaker.conf has a
+; higher priority than this
 ;
 ; See analog-output.conf.common for an explanation on the directives
 
 [General]
-priority = 100
+priority = 99
 
 [Element Hardware Master]
 switch = mute
@@ -37,18 +38,20 @@ override-map.2 = all-left,all-right
 switch = off
 volume = off
 
-; This profile path is intended to control the speaker, not the
+; This profile path is intended to control the default output, not the
 ; headphones. But it should not hurt if we leave the headphone jack
 ; enabled nonetheless.
 [Element Headphone]
 switch = mute
 volume = zero
 
+[Element Headphone2]
+switch = mute
+volume = zero
+
 [Element Speaker]
 switch = mute
-volume = merge
-override-map.1 = all
-override-map.2 = all-left,all-right
+volume = off
 
 [Element Front]
 switch = mute
diff --git a/src/modules/alsa/mixer/profile-sets/default.conf b/src/modules/alsa/mixer/profile-sets/default.conf
index ac41a8d3..046938fc 100644
--- a/src/modules/alsa/mixer/profile-sets/default.conf
+++ b/src/modules/alsa/mixer/profile-sets/default.conf
@@ -62,42 +62,42 @@ auto-profiles = yes
 [Mapping analog-mono]
 device-strings = hw:%f
 channel-map = mono
-paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
 paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line
 priority = 1
 
 [Mapping analog-stereo]
 device-strings = front:%f hw:%f
 channel-map = left,right
-paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
 paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line
 priority = 10
 
 [Mapping analog-surround-40]
 device-strings = surround40:%f
 channel-map = front-left,front-right,rear-left,rear-right
-paths-output = analog-output analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
 priority = 7
 direction = output
 
 [Mapping analog-surround-41]
 device-strings = surround41:%f
 channel-map = front-left,front-right,rear-left,rear-right,lfe
-paths-output = analog-output analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
 priority = 8
 direction = output
 
 [Mapping analog-surround-50]
 device-strings = surround50:%f
 channel-map = front-left,front-right,rear-left,rear-right,front-center
-paths-output = analog-output analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
 priority = 7
 direction = output
 
 [Mapping analog-surround-51]
 device-strings = surround51:%f
 channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe
-paths-output = analog-output analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
 priority = 8
 direction = output
 
@@ -105,7 +105,7 @@ direction = output
 device-strings = surround71:%f
 channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
 description = Analog Surround 7.1
-paths-output = analog-output analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
 priority = 7
 direction = output
 
diff --git a/src/modules/module-equalizer-sink.c b/src/modules/module-equalizer-sink.c
index 814a00fa..7c0ccd3a 100644
--- a/src/modules/module-equalizer-sink.c
+++ b/src/modules/module-equalizer-sink.c
@@ -412,7 +412,8 @@ static void dsp_logic(
     fftwf_complex * restrict output_window,//The transformed window'd src
     struct userdata *u){//Collection of constants
     const size_t overlap_size = PA_ROUND_UP(u->overlap_size, v_size);
-
+    float_vector_t x;
+    x.f[0] = x.f[1] = x.f[2] = x.f[3] = X;
 
     //assert(u->samples_gathered >= u->R);
     //use a linear-phase sliding STFT and overlap-add method
@@ -422,9 +423,8 @@ static void dsp_logic(
         float_vector_t *w = (float_vector_t*) (W + j);
         float_vector_t *s = (float_vector_t*) (src + j);
 //#if __SSE2__
-        d->m = _mm_mul_ps(w->m, s->m);
-//#else
-//        d->v = w->v * s->v;
+        d->m = _mm_mul_ps(x.m, _mm_mul_ps(w->m, s->m));
+//        d->v = x->v * w->v * s->v;
 //#endif
     }
     //zero padd the the remaining fft window
@@ -556,33 +556,41 @@ static void input_buffer(struct userdata *u, pa_memchunk *in){
 /* Called from I/O thread context */
 static int sink_input_pop_cb(pa_sink_input *i, size_t nbytes, pa_memchunk *chunk) {
     struct userdata *u;
-    size_t fs, target_samples;
-    struct timeval start, end;
+    size_t fs, target_samples, mbs;
+    //struct timeval start, end;
     pa_memchunk tchunk;
     pa_sink_input_assert_ref(i);
     pa_assert_se(u = i->userdata);
     pa_assert(chunk);
     pa_assert(u->sink);
     fs = pa_frame_size(&(u->sink->sample_spec));
+    nbytes = PA_MIN(nbytes, pa_mempool_block_size_max(u->sink->core->mempool));
     target_samples = PA_ROUND_UP(nbytes / fs, u->R);
+    mbs = pa_mempool_block_size_max(u->sink->core->mempool);
+    //pa_log_debug("vanilla mbs = %ld",mbs);
+    mbs = PA_ROUND_DOWN(mbs / fs, u->R);
+    mbs = PA_MAX(mbs, u->R);
+    target_samples = PA_MAX(target_samples, mbs);
+    //pa_log_debug("target samples: %ld", target_samples);
     if(u->first_iteration){
         //allocate request_size
         target_samples = PA_MAX(target_samples, u->window_size);
     }else{
         //allocate request_size + overlap
         target_samples += u->overlap_size;
-        alloc_input_buffers(u, target_samples);
     }
     alloc_input_buffers(u, target_samples);
+    //pa_log_debug("post target samples: %ld", target_samples);
     chunk->memblock = NULL;
 
     /* Hmm, process any rewind request that might be queued up */
     pa_sink_process_rewind(u->sink, 0);
 
     //pa_log_debug("start output-buffered %ld, input-buffered %ld, requested %ld",buffered_samples,u->samples_gathered,samples_requested);
-    pa_rtclock_get(&start);
+    //pa_rtclock_get(&start);
     do{
         size_t input_remaining = target_samples - u->samples_gathered;
+       // pa_log_debug("input remaining %ld samples", input_remaining);
         pa_assert(input_remaining > 0);
         while(pa_memblockq_peek(u->input_q, &tchunk) < 0){
             //pa_sink_render(u->sink, input_remaining * fs, &tchunk);
@@ -597,23 +605,23 @@ static int sink_input_pop_cb(pa_sink_input *i, size_t nbytes, pa_memchunk *chunk
         //pa_log_debug("asked for %ld input samples, got %ld samples",input_remaining,buffer->length/fs);
         /* copy new input */
         //pa_rtclock_get(start);
+       // pa_log_debug("buffering %ld bytes", tchunk.length);
         input_buffer(u, &tchunk);
         //pa_rtclock_get(&end);
         //pa_log_debug("Took %0.5f seconds to setup", pa_timeval_diff(end, start) / (double) PA_USEC_PER_SEC);
         pa_memblock_unref(tchunk.memblock);
     }while(u->samples_gathered < target_samples);
 
-    pa_rtclock_get(&end);
-    pa_log_debug("Took %0.6f seconds to get data", (double) pa_timeval_diff(&end, &start) / PA_USEC_PER_SEC);
+    //pa_rtclock_get(&end);
+    //pa_log_debug("Took %0.6f seconds to get data", (double) pa_timeval_diff(&end, &start) / PA_USEC_PER_SEC);
 
     pa_assert(u->fft_size >= u->window_size);
     pa_assert(u->R < u->window_size);
-    /* set the H filter */
-    pa_rtclock_get(&start);
+    //pa_rtclock_get(&start);
     /* process a block */
     process_samples(u, chunk);
-    pa_rtclock_get(&end);
-    pa_log_debug("Took %0.6f seconds to process", (double) pa_timeval_diff(&end, &start) / PA_USEC_PER_SEC);
+    //pa_rtclock_get(&end);
+    //pa_log_debug("Took %0.6f seconds to process", (double) pa_timeval_diff(&end, &start) / PA_USEC_PER_SEC);
 
     pa_assert(chunk->memblock);
     //pa_log_debug("gave %ld", chunk->length/fs);