1 files changed, 311 insertions, 0 deletions

diff --git a/akode/plugins/alsa_sink/alsa_sink.cpp b/akode/plugins/alsa_sink/alsa_sink.cpp
new file mode 100644
index 0000000..2abad2e
--- /dev/null
+++ b/akode/plugins/alsa_sink/alsa_sink.cpp
@@ -0,0 +1,311 @@
+/*  aKode: ALSA Sink
+
+    Copyright (C) 2004-2005 Allan Sandfeld Jensen <[email protected]>
+
+    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 <iostream>
+
+#include <alsa/asoundlib.h>
+#include <alsa/pcm.h>
+
+#include <audioframe.h>
+#include "alsa_sink.h"
+
+/*
+ * resume from suspend
+ */
+static int resume(snd_pcm_t *pcm)
+{
+  int res;
+  while ((res = snd_pcm_resume(pcm)) == -EAGAIN)
+    sleep(1);
+  if (! res)
+    return 0;
+  return snd_pcm_prepare(pcm);
+}
+
+namespace aKode {
+
+extern "C" { ALSASinkPlugin alsa_sink; }
+
+struct ALSASink::private_data
+{
+    private_data() : pcm_playback(0), buffer(0), error(false), can_pause(false) {};
+
+    snd_pcm_t *pcm_playback;
+
+    AudioConfiguration config;
+    int scale;
+    int filled, fragmentSize;
+    int sampleSize;
+    char* buffer;
+    bool error;
+    bool can_pause;
+};
+
+ALSASink::ALSASink()
+{
+    m_data = new private_data;
+}
+
+ALSASink::~ALSASink()
+{
+    close();
+    delete m_data;
+}
+
+bool ALSASink::open()
+{
+    int err = 0;
+    // open is non-blocking to make it possible to fail when occupied
+    err = snd_pcm_open(&m_data->pcm_playback, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+    if (err < 0) {
+        m_data->error = true;
+        return false;
+    }
+
+    // Set to blocking
+    snd_pcm_nonblock(m_data->pcm_playback, 0);
+
+    m_data->error = false;
+    return true;
+}
+
+void ALSASink::close()
+{
+    if (m_data->pcm_playback) {
+        snd_pcm_drain(m_data->pcm_playback);
+        snd_pcm_close(m_data->pcm_playback);
+    }
+    m_data->pcm_playback = 0;
+    m_data->error = false;
+}
+
+int ALSASink::setAudioConfiguration(const AudioConfiguration* config)
+{
+    if (m_data->error) return -1;
+
+    // Get back to OPEN state (is SETUP state enough with snd_pcm_drop?)
+    snd_pcm_state_t state = snd_pcm_state( m_data->pcm_playback );
+    if (state != SND_PCM_STATE_OPEN) {
+        close();
+        if (!open()) return -1;
+    }
+
+    int res = 0;
+    int wid = 1;
+    m_data->config = *config;
+    snd_pcm_hw_params_t *hw;
+    snd_pcm_hw_params_alloca(&hw);
+    snd_pcm_hw_params_any(m_data->pcm_playback, hw);
+    snd_pcm_hw_params_set_access(m_data->pcm_playback, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+    // Detect format:
+    snd_pcm_format_t format = SND_PCM_FORMAT_UNKNOWN;
+    // Test for float, 24 and 32 bit integer. Fall back to 16bit
+    if (m_data->config.sample_width<0) {
+        if (snd_pcm_hw_params_test_format(m_data->pcm_playback, hw,  SND_PCM_FORMAT_FLOAT ) == 0) {
+            format = SND_PCM_FORMAT_FLOAT;
+            m_data->scale = 1;
+            wid = 4;
+            goto found_format;
+        }
+        // Try 16bit then
+        m_data->config.sample_width = 16;
+        res = 1;
+    }
+    if (m_data->config.sample_width > 24 && m_data->config.sample_width <=32) {
+        if (snd_pcm_hw_params_test_format(m_data->pcm_playback, hw, SND_PCM_FORMAT_S32) == 0) {
+            format = SND_PCM_FORMAT_S32;
+            m_data->scale = 1<<(32-config->sample_width);
+            wid = 4;
+            goto found_format;
+        }
+        // Try 24bit then
+        m_data->config.sample_width = 24;
+        res = 1;
+    }
+    if (m_data->config.sample_width > 16 && m_data->config.sample_width <= 24) {
+        if (snd_pcm_hw_params_test_format(m_data->pcm_playback, hw,  SND_PCM_FORMAT_S24 ) == 0) {
+            format = SND_PCM_FORMAT_S24;
+            m_data->scale = 1<<(24-config->sample_width);
+            wid = 4;
+            goto found_format;
+        }
+        // Try 16bit then
+        m_data->config.sample_width = 16;
+    }
+    // If the driver doesnt support 8 or 16 bit, we will fail completly
+    if (m_data->config.sample_width<=8) {
+        format = SND_PCM_FORMAT_S8;
+        m_data->scale = 1<<(8-config->sample_width);
+        wid = 1;
+        goto found_format;
+    }
+    else
+    if (m_data->config.sample_width<=16) {
+        format = SND_PCM_FORMAT_S16;
+        m_data->scale = 1<<(16-config->sample_width);
+        wid = 2;
+        goto found_format;
+    }
+
+found_format:
+    if (format != SND_PCM_FORMAT_UNKNOWN)
+        snd_pcm_hw_params_set_format(m_data->pcm_playback, hw, format);
+    else
+        return -1;
+
+    unsigned int rate = config->sample_rate;
+    snd_pcm_hw_params_set_rate_near(m_data->pcm_playback, hw, &rate, 0);
+    if (m_data->config.sample_rate != rate) {
+        m_data->config.sample_rate = rate;
+        res = 1;
+    }
+
+    snd_pcm_hw_params_set_channels(m_data->pcm_playback, hw, config->channels);
+
+
+    m_data->fragmentSize = 1024;
+    snd_pcm_uframes_t period_size = m_data->fragmentSize / (wid*config->channels);
+    snd_pcm_hw_params_set_period_size_near(m_data->pcm_playback, hw, &period_size, 0);
+
+    m_data->fragmentSize = period_size * (wid*config->channels);
+//     std::cerr << "akode: ALSA fragment-size: " << m_data->fragmentSize << "\n";
+
+    delete m_data->buffer;
+    m_data->buffer = new char [m_data->fragmentSize];
+    m_data->filled = 0;
+
+    if (snd_pcm_hw_params(m_data->pcm_playback, hw) < 0) {
+        return -1;
+    }
+    else {
+        m_data->can_pause = (snd_pcm_hw_params_can_pause(hw) == 1);
+        return res;
+    }
+}
+
+const AudioConfiguration* ALSASink::audioConfiguration() const
+{
+    return &m_data->config;
+}
+
+template<class T>
+bool ALSASink::_writeFrame(AudioFrame* frame)
+{
+    int channels = m_data->config.channels;
+
+    long i = 0;
+    T* buffer = (T*)m_data->buffer;
+    T** data = (T**)frame->data;
+    while(true) {
+        if (m_data->filled >= m_data->fragmentSize)
+        xrun:
+        {
+            int frames = snd_pcm_bytes_to_frames(m_data->pcm_playback, m_data->filled);
+            int status = snd_pcm_writei(m_data->pcm_playback, m_data->buffer, frames);
+            if (status == -EPIPE) {
+                snd_pcm_prepare(m_data->pcm_playback);
+                //std::cerr << "akode: ALSA xrun\n";
+                goto xrun;
+            }
+            else if (status < 0) return false;
+            int bytes = snd_pcm_frames_to_bytes(m_data->pcm_playback, status);
+            if (m_data->filled != bytes) {
+                int rest = m_data->filled - bytes;
+                //std::cerr << "akode: ALSA write-remainder: " << rest << "\n";
+                memmove(m_data->buffer, m_data->buffer + bytes, rest);
+                m_data->filled = rest;
+            } else
+                m_data->filled = 0;
+
+        }
+        if (i >= frame->length) break;
+        for(int j=0; j<channels; j++) {
+            buffer[m_data->filled/sizeof(T)] = (data[j][i])*m_data->scale;
+            m_data->filled+=sizeof(T);
+        }
+        i++;
+    }
+
+    if (snd_pcm_state( m_data->pcm_playback ) == SND_PCM_STATE_PREPARED)
+        snd_pcm_start(m_data->pcm_playback);
+
+    return true;
+}
+
+bool ALSASink::writeFrame(AudioFrame* frame)
+{
+    if (m_data->error) return false;
+    if (!frame) return false;
+
+    if ( frame->sample_width != m_data->config.sample_width
+      || frame->channels != m_data->config.channels
+      || frame->sample_rate != m_data->config.sample_rate)
+    {
+        if (setAudioConfiguration(frame) < 0)
+            return false;
+    }
+
+    if ( snd_pcm_state(m_data->pcm_playback) == SND_PCM_STATE_SUSPENDED ) {
+      int res = ::resume(m_data->pcm_playback);
+      if (res < 0)
+        return false;
+    }
+    else
+    if (snd_pcm_state( m_data->pcm_playback ) == SND_PCM_STATE_PAUSED)
+        snd_pcm_pause(m_data->pcm_playback, 0);
+
+    if (snd_pcm_state( m_data->pcm_playback ) == SND_PCM_STATE_SETUP)
+        snd_pcm_prepare(m_data->pcm_playback);
+
+    if (frame->sample_width<0)
+        return _writeFrame<float>(frame);
+    else
+    if (frame->sample_width<=8)
+        return _writeFrame<int8_t>(frame);
+    else
+    if (frame->sample_width<=16)
+        return _writeFrame<int16_t>(frame);
+    else
+    if (frame->sample_width<=32)
+        return _writeFrame<int32_t>(frame);
+
+    return false;
+}
+
+void ALSASink::pause()
+{
+    if (m_data->error) return;
+
+    if (m_data->can_pause) {
+        snd_pcm_pause(m_data->pcm_playback, 1);
+    }
+
+}
+
+// Do not confuse this with snd_pcm_resume which is used to resume from a suspend
+void ALSASink::resume()
+{
+    if (m_data->error) return;
+
+    if (snd_pcm_state( m_data->pcm_playback ) == SND_PCM_STATE_PAUSED)
+        snd_pcm_pause(m_data->pcm_playback, 0);
+}
+
+} // namespace