added multichannel support

git-svn-id: http://svn.code.sf.net/p/qmmp-dev/code/trunk/qmmp@4530 90c681e8-e032-0410-971d-27865f9a5e38

10 files changed, 155 insertions, 39 deletions

diff --git a/src/plugins/Effect/Effect.pro b/src/plugins/Effect/Effect.pro
index 336647a4b..b073f08f3 100644
--- a/src/plugins/Effect/Effect.pro
+++ b/src/plugins/Effect/Effect.pro
@@ -1,15 +1,15 @@
 include (../../../qmmp.pri)
 TEMPLATE = subdirs
 
-SUBDIRS += crossfade stereo
+#SUBDIRS += crossfade stereo
 
 contains(CONFIG, BS2B_PLUGIN){
-SUBDIRS += bs2b
+#SUBDIRS += bs2b
 }
 
 unix {
     SUBDIRS += srconverter
     contains(CONFIG, LADSPA_PLUGIN){
-    SUBDIRS += ladspa
+#    SUBDIRS += ladspa
     }
 }
diff --git a/src/plugins/Effect/srconverter/srconverter.cpp b/src/plugins/Effect/srconverter/srconverter.cpp
index fcd77bc04..324dbb051 100644
--- a/src/plugins/Effect/srconverter/srconverter.cpp
+++ b/src/plugins/Effect/srconverter/srconverter.cpp
@@ -1,5 +1,5 @@
 /***************************************************************************
- *   Copyright (C) 2007-2013 by Ilya Kotov                                 *
+ *   Copyright (C) 2007-2014 by Ilya Kotov                                 *
  *   forkotov02@hotmail.ru                                                 *
  *                                                                         *
  *   This program is free software; you can redistribute it and/or modify  *
@@ -115,12 +115,12 @@ void SRConverter::applyEffect(Buffer *b)
     }
 }
 
-void SRConverter::configure(quint32 freq, int chan,  Qmmp::AudioFormat format)
+void SRConverter::configure(quint32 freq, ChannelMap map,  Qmmp::AudioFormat format)
 {
     freeSRC();
     if(freq != m_overSamplingFs && format != Qmmp::PCM_S8)
     {
-        m_src_state = src_new(m_converter_type, chan, &m_srcError);
+        m_src_state = src_new(m_converter_type, map.count(), &m_srcError);
         if (m_src_state)
         {
             m_src_data.src_ratio = (float)m_overSamplingFs/(float)freq;
@@ -129,7 +129,7 @@ void SRConverter::configure(quint32 freq, int chan,  Qmmp::AudioFormat format)
         else
             qDebug("SRConverter: src_new(): %s", src_strerror(m_srcError));
     }
-    Effect::configure(m_overSamplingFs, chan, format);
+    Effect::configure(m_overSamplingFs, map, format);
     m_sz = audioParameters().sampleSize();
 }
 
diff --git a/src/plugins/Effect/srconverter/srconverter.h b/src/plugins/Effect/srconverter/srconverter.h
index f25ebc491..b214c866f 100644
--- a/src/plugins/Effect/srconverter/srconverter.h
+++ b/src/plugins/Effect/srconverter/srconverter.h
@@ -1,5 +1,5 @@
 /***************************************************************************
- *   Copyright (C) 2007-2013 by Ilya Kotov                                 *
+ *   Copyright (C) 2007-2014 by Ilya Kotov                                 *
  *   forkotov02@hotmail.ru                                                 *
  *                                                                         *
  *   This program is free software; you can redistribute it and/or modify  *
@@ -34,7 +34,7 @@ public:
     virtual ~SRConverter();
 
     void applyEffect(Buffer *b);
-    void configure(quint32 freq, int chan, Qmmp::AudioFormat format);
+    void configure(quint32 freq, ChannelMap map, Qmmp::AudioFormat format);
 
 private:
     void freeSRC();
diff --git a/src/plugins/Input/Input.pro b/src/plugins/Input/Input.pro
index 9b00864e6..0149aa949 100644
--- a/src/plugins/Input/Input.pro
+++ b/src/plugins/Input/Input.pro
@@ -1,53 +1,53 @@
 include(../../../qmmp.pri)
 
-SUBDIRS += mad cue vorbis sndfile wavpack
 TEMPLATE = subdirs
+SUBDIRS += mad #cue vorbis sndfile wavpack
 
 contains(CONFIG, FLAC_PLUGIN){
     SUBDIRS += flac
 }
 
 contains(CONFIG, MUSEPACK_PLUGIN){
-    SUBDIRS += mpc
+#    SUBDIRS += mpc
 }
 
 contains(CONFIG, MODPLUG_PLUGIN){
-    SUBDIRS += modplug
+#    SUBDIRS += modplug
 }
 
 contains(CONFIG, FFMPEG_PLUGIN){
-   contains(CONFIG, FFMPEG_LEGACY){
-       SUBDIRS += ffmpeg_legacy
-    }else{
-       SUBDIRS += ffmpeg
-    }
+#   contains(CONFIG, FFMPEG_LEGACY){
+#       SUBDIRS += ffmpeg_legacy
+#    }else{
+#       SUBDIRS += ffmpeg
+#    }
 }
 
 contains(CONFIG, GME_PLUGIN){
-  SUBDIRS += gme
+#  SUBDIRS += gme
 }
 
 contains(CONFIG, OPUS_PLUGIN){
-  SUBDIRS += opus
+#  SUBDIRS += opus
 }
 
 contains(CONFIG, CDAUDIO_PLUGIN){
-  SUBDIRS += cdaudio
+#  SUBDIRS += cdaudio
 }
 
 contains(CONFIG, SID_PLUGIN){
-  SUBDIRS += sid
+#  SUBDIRS += sid
 }
 
 unix{
 
 contains(CONFIG, AAC_PLUGIN){
-  SUBDIRS += aac
+#  SUBDIRS += aac
 }
 
 
 contains(CONFIG, WILDMIDI_PLUGIN){
-  SUBDIRS += wildmidi
+#  SUBDIRS += wildmidi
 }
 
 
diff --git a/src/plugins/Input/flac/decoder_flac.cpp b/src/plugins/Input/flac/decoder_flac.cpp
index acd76f159..4dfca6839 100644
--- a/src/plugins/Input/flac/decoder_flac.cpp
+++ b/src/plugins/Input/flac/decoder_flac.cpp
@@ -393,17 +393,25 @@ bool DecoderFLAC::initialize()
         data()->ok = 0;
         return false;
     }
+    ChannelMap channel_map = findChannelMap(data()->channels);
+
+    if(channel_map.isEmpty())
+    {
+        qWarning("DecoderFLAC: unsupported number of channels: %d", data()->channels);
+        return false;
+    }
+
     switch(data()->bits_per_sample)
     {
     case 8:
-        configure(data()->sample_rate, data()->channels, Qmmp::PCM_S8);
+        configure(data()->sample_rate, channel_map, Qmmp::PCM_S8);
         break;
     case 16:
-        configure(data()->sample_rate, data()->channels, Qmmp::PCM_S16LE);
+        configure(data()->sample_rate, channel_map, Qmmp::PCM_S16LE);
         break;
     case 24:
     case 32:
-        configure(data()->sample_rate, data()->channels, Qmmp::PCM_S32LE);
+        configure(data()->sample_rate, channel_map, Qmmp::PCM_S32LE);
         break;
     default:
         return false;
@@ -555,3 +563,66 @@ uint DecoderFLAC::findID3v2(char *data, ulong size) //retuns ID3v2 tag size
     }
     return 0;
 }
+
+ChannelMap DecoderFLAC::findChannelMap(int channels)
+{
+    ChannelMap map;
+    switch (channels)
+    {
+    case 1:
+        map << Qmmp::CHAN_FRONT_LEFT;
+        break;
+    case 2:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT;
+        break;
+    case 3:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_FRONT_CENTER;
+        break;
+    case 4:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_REAR_LEFT
+            << Qmmp::CHAN_REAR_RIGHT;
+        break;
+    case 5:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_FRONT_CENTER
+            << Qmmp::CHAN_REAR_LEFT
+            << Qmmp::CHAN_REAR_RIGHT;
+        break;
+    case 6:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_FRONT_CENTER
+            << Qmmp::CHAN_LFE
+            << Qmmp::CHAN_REAR_LEFT
+            << Qmmp::CHAN_REAR_RIGHT;
+        break;
+    case 7:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_FRONT_CENTER
+            << Qmmp::CHAN_LFE
+            << Qmmp::CHAN_REAR_CENTER
+            << Qmmp::CHAN_SIDE_LEFT
+            << Qmmp::CHAN_SIDE_RIGHT;
+        break;
+    case 8:
+        map << Qmmp::CHAN_FRONT_LEFT
+            << Qmmp::CHAN_FRONT_RIGHT
+            << Qmmp::CHAN_FRONT_CENTER
+            << Qmmp::CHAN_LFE
+            << Qmmp::CHAN_REAR_LEFT
+            << Qmmp::CHAN_REAR_RIGHT
+            << Qmmp::CHAN_SIDE_LEFT
+            << Qmmp::CHAN_SIDE_RIGHT;
+        break;
+    default:
+        ;
+    }
+    return map;
+}
diff --git a/src/plugins/Input/flac/decoder_flac.h b/src/plugins/Input/flac/decoder_flac.h
index e64e376eb..4b4a2f668 100644
--- a/src/plugins/Input/flac/decoder_flac.h
+++ b/src/plugins/Input/flac/decoder_flac.h
@@ -82,6 +82,7 @@ private:
     // helper functions
     void deinit();
     uint findID3v2(char *data, ulong size); //retuns ID3v2 tag size
+    ChannelMap findChannelMap(int channels);
 
     struct flac_data *m_data;
     qint64 length_in_bytes;
diff --git a/src/plugins/Input/mad/decoder_mad.cpp b/src/plugins/Input/mad/decoder_mad.cpp
index b4087c06b..08b2486fb 100644
--- a/src/plugins/Input/mad/decoder_mad.cpp
+++ b/src/plugins/Input/mad/decoder_mad.cpp
@@ -120,7 +120,12 @@ bool DecoderMAD::initialize()
     mad_frame_mute (&m_frame);
     m_stream.next_frame = 0;
     m_stream.sync = 0;
-    configure(m_freq, m_channels, Qmmp::PCM_S16LE);
+    ChannelMap map;
+    if(m_channels == 1)
+        map << Qmmp::CHAN_FRONT_LEFT;
+    else
+        map << Qmmp::CHAN_FRONT_LEFT << Qmmp::CHAN_FRONT_RIGHT;
+    configure(m_freq, map, Qmmp::PCM_S16LE);
     m_inited = true;
     return true;
 }
diff --git a/src/plugins/Output/Output.pro b/src/plugins/Output/Output.pro
index d07513257..7dc1f0ddb 100644
--- a/src/plugins/Output/Output.pro
+++ b/src/plugins/Output/Output.pro
@@ -4,20 +4,20 @@ TEMPLATE = subdirs
 win32:SUBDIRS += waveout
 win32:SUBDIRS += directsound
 
-SUBDIRS += null
+#SUBDIRS += null
 
 unix{
 
 contains(CONFIG, JACK_PLUGIN){
-    SUBDIRS += jack
+#    SUBDIRS += jack
 }
 
 contains(CONFIG, OSS_PLUGIN){
-    SUBDIRS += oss
+#    SUBDIRS += oss
 }
 
 contains(CONFIG, PULSE_AUDIO_PLUGIN){
-    SUBDIRS += pulseaudio
+#    SUBDIRS += pulseaudio
 }
 
 contains(CONFIG, ALSA_PLUGIN){
@@ -25,7 +25,7 @@ contains(CONFIG, ALSA_PLUGIN){
 }
 
 contains(CONFIG, OSS4_PLUGIN){
-    SUBDIRS += oss4
+#    SUBDIRS += oss4
 }
 
 }
diff --git a/src/plugins/Output/alsa/outputalsa.cpp b/src/plugins/Output/alsa/outputalsa.cpp
index 5774579f5..f12f02060 100644
--- a/src/plugins/Output/alsa/outputalsa.cpp
+++ b/src/plugins/Output/alsa/outputalsa.cpp
@@ -44,6 +44,18 @@ OutputALSA::OutputALSA() : m_inited(false)
     m_prebuf_size = 0;
     m_prebuf_fill = 0;
     m_can_pause = false;
+
+    m_alsa_channels[SND_CHMAP_NA] =   Qmmp::CHAN_NULL;
+    m_alsa_channels[SND_CHMAP_MONO] = Qmmp::CHAN_FRONT_CENTER;
+    m_alsa_channels[SND_CHMAP_FL]   = Qmmp::CHAN_FRONT_LEFT;
+    m_alsa_channels[SND_CHMAP_FR]   = Qmmp::CHAN_FRONT_RIGHT;
+    m_alsa_channels[SND_CHMAP_RL]   = Qmmp::CHAN_REAR_LEFT;
+    m_alsa_channels[SND_CHMAP_RR]   = Qmmp::CHAN_REAR_RIGHT;
+    m_alsa_channels[SND_CHMAP_FC]   = Qmmp::CHAN_FRONT_CENTER;
+    m_alsa_channels[SND_CHMAP_LFE]  = Qmmp::CHAN_LFE;
+    m_alsa_channels[SND_CHMAP_SL]   = Qmmp::CHAN_SIDE_LEFT;
+    m_alsa_channels[SND_CHMAP_SR]   = Qmmp::CHAN_SIDE_RIGHT;
+    m_alsa_channels[SND_CHMAP_RC]   = Qmmp::CHAN_REAR_CENTER;
 }
 
 OutputALSA::~OutputALSA()
@@ -52,7 +64,7 @@ OutputALSA::~OutputALSA()
     free (pcm_name);
 }
 
-bool OutputALSA::initialize(quint32 freq, int chan, Qmmp::AudioFormat format)
+bool OutputALSA::initialize(quint32 freq, ChannelMap map, Qmmp::AudioFormat format)
 {
     m_inited = false;
 
@@ -141,12 +153,17 @@ bool OutputALSA::initialize(quint32 freq, int chan, Qmmp::AudioFormat format)
         qWarning("OutputALSA: The rate %d Hz is not supported by your hardware.\n==> Using %d Hz instead.", rate, exact_rate);
         rate = exact_rate;
     }
-    uint c = chan;
+    uint c = map.count();
     if ((err = snd_pcm_hw_params_set_channels_near(pcm_handle, hwparams, &c)) < 0)
     {
         qWarning("OutputALSA: Error setting channels: %s", snd_strerror(err));
         return false;
     }
+    if (c != (uint)map.count())
+    {
+        qWarning("OutputALSA: The channel number %d is not supported by your hardware", map.count());
+        qWarning("==> Using %d instead.", c);
+    }
     if ((err = snd_pcm_hw_params_set_period_time_near(pcm_handle, hwparams, &period_time ,0)) < 0)
     {
         qWarning("OutputALSA: Error setting period time: %s", snd_strerror(err));
@@ -190,11 +207,30 @@ bool OutputALSA::initialize(quint32 freq, int chan, Qmmp::AudioFormat format)
     m_chunk_size = period_size;
     m_can_pause = snd_pcm_hw_params_can_pause(hwparams) && use_pause;
     qDebug("OutputALSA: can pause: %d", m_can_pause);
-    configure(rate, chan, format); //apply configuration
+
+    //channel map configuration
+    snd_pcm_chmap_t *chmap = snd_pcm_get_chmap(pcm_handle);
+    if(!chmap)
+    {
+        qWarning("OutputALSA: Unable to receive current channel map: %s", snd_strerror(err));
+        return false;
+    }
+    char tmp[256];
+    memset(tmp,0,256);
+    snd_pcm_chmap_print(chmap, 256, tmp);
+    qDebug("OutputALSA: received channel map: %s",tmp);
+    ChannelMap out_map;
+    for(uint i = 0; i < chmap->channels; ++i)
+    {
+        if(m_alsa_channels.keys().contains(chmap->pos[i]))
+            out_map.append(m_alsa_channels.value(chmap->pos[i]));
+        else
+            out_map.append(Qmmp::CHAN_NULL);
+    }
+    configure(exact_rate, out_map, format); //apply configuration
     //create alsa prebuffer;
     m_prebuf_size = 2 * snd_pcm_frames_to_bytes(pcm_handle, m_chunk_size); //buffer for two periods
     m_prebuf = (uchar *)malloc(m_prebuf_size);
-
     m_inited = true;
     return true;
 }
diff --git a/src/plugins/Output/alsa/outputalsa.h b/src/plugins/Output/alsa/outputalsa.h
index 1600e3b29..d378b813c 100644
--- a/src/plugins/Output/alsa/outputalsa.h
+++ b/src/plugins/Output/alsa/outputalsa.h
@@ -18,8 +18,8 @@
  *   51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.         *
  ***************************************************************************/
 
-#ifndef   OUTPUTALSA_H
-#define   OUTPUTALSA_H
+#ifndef OUTPUTALSA_H
+#define OUTPUTALSA_H
 
 class OutputALSA;
 
@@ -28,6 +28,7 @@ extern "C"
 #include <alsa/asoundlib.h>
 }
 
+#include <QHash>
 #include <qmmp/output.h>
 #include <qmmp/volume.h>
 
@@ -38,7 +39,7 @@ public:
     OutputALSA();
     ~OutputALSA();
 
-    bool initialize(quint32, int, Qmmp::AudioFormat format);
+    bool initialize(quint32, ChannelMap map, Qmmp::AudioFormat format);
     //output api
     qint64 latency();
     qint64 writeAudio(unsigned char *data, qint64 maxSize);
@@ -63,6 +64,8 @@ private:
     qint64 m_prebuf_size;
     qint64 m_prebuf_fill;
     bool m_can_pause;
+    //channel conversions
+    QHash <quint16, Qmmp::ChannelPosition> m_alsa_channels;
 };
 
 class VolumeALSA : public Volume