1 files changed, 227 insertions, 0 deletions
diff --git a/akode/plugins/jack_sink/jack_sink.cpp b/akode/plugins/jack_sink/jack_sink.cpp
new file mode 100644
index 0000000..4824165
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+++ b/akode/plugins/jack_sink/jack_sink.cpp
@@ -0,0 +1,227 @@
+/*  aKode: JACK-Sink
+
+    Copyright (C) 2004 Allan Sandfeld Jensen <kde@carewolf.com>
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Library General Public
+    License as published by the Free Software Foundation; either
+    version 2 of the License, or (at your option) any later version.
+
+    This library 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
+    Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public License
+    along with this library; see the file COPYING.LIB.  If not, write to
+    the Free Software Foundation, Inc., 51 Franklin Steet, Fifth Floor,
+    Boston, MA 02110-1301, USA.
+*/
+
+#include "config.h"
+
+#ifdef HAVE_STDINT_H
+#include <stdint.h>
+#elif HAVE_INTTYPES_H
+#include <inttypes.h>
+#endif
+#include <jack/jack.h>
+
+#include <audioframe.h>
+#include <audiobuffer.h>
+#include "jack_sink.h"
+
+#include <iostream>
+
+namespace aKode {
+
+extern "C" { JACKSinkPlugin jack_sink; }
+
+struct JACKSink::private_data
+{
+    private_data() : left_port(0), right_port(0), client(0), error(false), sample_rate(0), buffer(8),pos(0) {};
+//    jack_port_t *input_port;
+    jack_port_t *left_port;
+    jack_port_t *right_port;
+
+    jack_client_t *client;
+
+    bool error;
+    unsigned int sample_rate;
+
+    AudioConfiguration config;
+    AudioBuffer buffer;
+    AudioFrame current;
+    int pos;
+};
+
+static int process (jack_nframes_t nframes, void *arg)
+{
+    JACKSink::private_data *m_data = (JACKSink::private_data*)arg;
+
+    jack_default_audio_sample_t *out1=0, *out2=0;
+
+    if (m_data->left_port)
+        out1 = (jack_default_audio_sample_t *) jack_port_get_buffer (m_data->left_port, nframes);
+    if (m_data->right_port)
+        out2 = (jack_default_audio_sample_t *) jack_port_get_buffer (m_data->right_port, nframes);
+
+    if (!out2 && !out1) return 0;
+
+    unsigned int n = 0;
+    float** buffer = (float**)m_data->current.data;
+    while(n < nframes) {
+        if (m_data->pos >= m_data->current.length) {
+            if (!m_data->buffer.get(&m_data->current, false)) break;
+            m_data->pos = 0;
+            buffer = (float**)m_data->current.data;
+        }
+
+        if (out1) out1[n] = buffer[0][m_data->pos];
+        if (out2) out2[n] = buffer[1][m_data->pos];
+
+        n++;
+        m_data->pos++;
+    }
+
+    return n;
+}
+
+static void shutdown (void *arg)
+{
+    JACKSink::private_data *m_data = (JACKSink::private_data*)arg;
+
+    // Jack has shut down, so the sink is in fatal error.
+    m_data->error = true;
+    m_data->buffer.release();
+}
+
+JACKSink::JACKSink()
+{
+    m_data = new private_data;
+}
+
+bool JACKSink::open() {
+    m_data->client = jack_client_new("akode_client");
+    if (!m_data->client) {
+        m_data->error = true;
+        return false;
+    }
+    jack_set_process_callback(m_data->client, process, (void*)m_data);
+    jack_on_shutdown(m_data->client, shutdown, (void*)m_data);
+
+    m_data->sample_rate = jack_get_sample_rate(m_data->client);
+//    jack_set_sample_rate_callback(m_data->client, sample_rate, (void*)m_data);
+
+    if (jack_activate(m_data->client)) {
+        m_data->error = true;
+        std::cout << "aKode::Jack: Activate failed\n";
+        return false;
+    }
+    const char** names = jack_get_ports (m_data->client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
+    while (*names) {
+        std::cout << *names << std::endl;
+        names++;
+    }
+
+    return true;
+}
+
+JACKSink::~JACKSink()
+{
+    if (m_data->left_port)
+        jack_port_unregister(m_data->client, m_data->left_port);
+    if (m_data->right_port)
+        jack_port_unregister(m_data->client, m_data->right_port);
+    if (m_data->client)
+        jack_deactivate(m_data->client);
+    delete m_data;
+}
+
+int JACKSink::setAudioConfiguration(const AudioConfiguration* config)
+{
+    if (m_data->error) return -1;
+
+    int res = 0;
+    m_data->config = *config;
+
+    if (config->channel_config != MonoStereo ) return -1;
+    m_data->left_port = jack_port_register (m_data->client, "left", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (jack_connect (m_data->client, jack_port_name (m_data->left_port), "alsa_pcm:playback_1")) {
+        m_data->error = true;
+        return -1;
+    }
+    if (config->channels > 1) {
+        m_data->right_port = jack_port_register (m_data->client, "right", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+        if (jack_connect (m_data->client, jack_port_name (m_data->right_port), "alsa_pcm:playback_2")) {
+            m_data->config.channels = 1;
+            res = 1;
+        }
+    }
+
+    if (config->sample_width != -32) {
+        res = 1;
+        m_data->config.sample_width = -32;
+    }
+
+    if (config->sample_rate != m_data->sample_rate) {
+        res = 1;
+        m_data->config.sample_rate = m_data->sample_rate;
+    }
+
+    return res;
+}
+
+const AudioConfiguration* JACKSink::audioConfiguration() const
+{
+    return &m_data->config;
+}
+
+template<typename S>
+void JACKSink::convertFrame(AudioFrame *in, AudioFrame *out)
+{
+    float scale = (float)(1<<(in->sample_width-1));
+    scale = 1.0/scale;
+//    float scale = 1;
+
+    out->reserveSpace(&m_data->config, in->length);
+
+    int channels = in->channels;
+    S** indata = (S**)in->data;
+    float** outdata = (float**)out->data;
+    for(int j=0; j<in->length; j++) {
+        for(int i=0; i<channels; i++) {
+            outdata[i][j] = scale*indata[i][j];
+        }
+    }
+}
+
+bool JACKSink::writeFrame(AudioFrame* frame)
+{
+    if ( m_data->error ) return false;
+
+    if ( frame->channels != m_data->config.channels )
+    {
+        if (setAudioConfiguration(frame)!=0)
+            return false;
+    }
+    if (frame->length == 0) return true;
+
+    // this shouldn't really happen
+    if (frame->sample_width>0) {
+        AudioFrame out;
+        if (frame->sample_width<=8)
+            convertFrame<int8_t>(frame, &out);
+        else
+        if (frame->sample_width<=16)
+            convertFrame<int16_t>(frame, &out);
+        else
+        if (frame->sample_width<=32)
+            convertFrame<int32_t>(frame, &out);
+        return m_data->buffer.put(&out, true);
+    } else
+        return m_data->buffer.put(frame, true);
+
+}
+
+} // namespace