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coreaudio_example.cc

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+/*
+CoreAudio, or how to actually use the worst documented audio API in history.
+
+Recently I started working on a CoreAudio plugin for audacious, to replace the
+old one which was removed in Audacious 3.2, since the mac port was abandoned
+due to the fact that Gtk+ is horrible on mac. Instead of updating the old
+CoreAudio plugin which was very limited and consisted of bad code ported from
+the XMMS days, I decided to start from scratch, using the simple SDL output
+plugin as a model.
+
+The API itself is okay, but the documentation is misleading. For example, they
+encourage you to set up an Audio Unit Graph for simple audio playback with
+audio control. At least on OS X, this isn't really necessary (thankfully).
+Since this isn't necessary, I will show you how to do it the easy way, since
+everyone seems to like to over-complicate things.
+
+These examples assume you have a 'pump thread' which is pumping audio to a
+circular buffer. Implementing that is not covered here, but isn't really very
+hard to do.
+*/
+
+/*
+This is the API we will be implementing, it is pretty straight-forward. There
+are 6 functions which are provided:
+
+init: start up the audio system
+cleanup: shut down the audio system
+set_volume: set the volume
+open_audio: set up a stream for playback
+close_audio: shuts down the playback code
+pause_audio: pauses the playback code
+
+The API will handle format conversion between your playback code's preferred
+format type and CoreAudio's native format -- float32, non-interleaved linear
+pcm. We will do this using a lookup table based on the enum...
+*/
+
+/* CoreAudio utility functions, public domain.
+   http://kaniini.dereferenced.org/2014/08/31/CoreAudio-sucks.html */
+
+#include "coreaudio_example.h"
+
+namespace coreaudio_example {
+
+struct CoreAudioFormatDescriptionMap {
+    enum format_type type;
+    int bits_per_sample;
+    int bytes_per_sample;
+    unsigned int flags;
+};
+static struct CoreAudioFormatDescriptionMap format_map[] = {
+    {FMT_S16_LE, 16, sizeof (int16_t), kAudioFormatFlagIsSignedInteger},
+    {FMT_S16_BE, 16, sizeof (int16_t), kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian},
+    {FMT_S32_LE, 32, sizeof (int32_t), kAudioFormatFlagIsSignedInteger},
+    {FMT_S32_BE, 32, sizeof (int32_t), kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsBigEndian},
+    {FMT_FLOAT,  32, sizeof (float),   kAudioFormatFlagIsFloat},
+};
+
+/*
+This is our lookup table which handles looking up the format specification
+attributes. There are a few members in each entry which are interesting:
+
+- type: this is our format type (i.e. FMT_S16_LE) which we will be looking up
+- bits_per_sample: this is the number of bits of data that should be included
+  in a sample. CoreAudio takes the lower bits of data, not the higher bits of
+  data. So if you were to implement 24-bit PCM support, you would pack the data
+  into a series of 32-bit integers and tell CoreAudio to use 24 bits of each
+  sample for PCM data.
+- bytes_per_sample: this is the size in bytes of the type which contains each
+  sample.
+- flags: This is a bitfield of many flags, the ones mentioned here are:
+- kAudioFormatIsSignedInteger: The audio format is a signed integer. If this is
+  not set, CoreAudio assumes it's unsigned.
+- kAudioFormatIsBigEndian: The audio format is in big-endian notation. If this
+  is not set, CoreAudio assumes it's little-endian.
+- kAudioFormatIsFloat: The audio format is in 32-bit floating point format.
+
+We now continue with actually initializing our output unit so we can use it.
+*/
+
+static AudioComponent output_comp;
+static AudioComponentInstance output_instance;
+
+bool init (void)
+{
+    /* open the default audio device */
+    AudioComponentDescription desc;
+    desc.componentType = kAudioUnitType_Output;
+    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+    desc.componentFlags = 0;
+    desc.componentFlagsMask = 0;
+    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+
+    output_comp = AudioComponentFindNext (nullptr, & desc);
+    if (! output_comp)
+    {
+        fprintf (stderr, "Failed to open default audio device.\n");
+        return false;
+    }
+
+    if (AudioComponentInstanceNew (output_comp, & output_instance))
+    {
+        fprintf (stderr, "Failed to open default audio device.\n");
+        return false;
+    }
+
+    return true;
+}
+
+void cleanup (void)
+{
+    AudioUnitInstanceDispose (output_instance);
+}
+
+/*
+The init () and cleanup () routines handle bringing up CoreAudio in the app.
+This gives you an output unit you can send data to using a callback. Now we
+should actually set up the unit for playback...
+*/
+
+bool open_audio (int format, int rate, int chan, AURenderCallbackStruct * callback)
+{
+    struct CoreAudioFormatDescriptionMap * m = nullptr;
+
+    for (struct CoreAudioFormatDescriptionMap it : format_map)
+    {
+        if (it.type == format)
+        {
+            m = & it;
+            break;
+        }
+    }
+
+    if (! m)
+    {
+        fprintf (stderr, "The requested audio format %d is unsupported.\n", format);
+        return false;
+    }
+
+    if (AudioUnitInitialize (output_instance))
+    {
+        fprintf (stderr, "Unable to initialize audio unit instance\n");
+        return false;
+    }
+
+    AudioStreamBasicDescription streamFormat;
+    streamFormat.mSampleRate = rate;
+    streamFormat.mFormatID = kAudioFormatLinearPCM;
+    streamFormat.mFormatFlags = m->mFormatFlags;
+    streamFormat.mFramesPerPacket = 1;
+    streamFormat.mChannelsPerFrame = chan;
+    streamFormat.mBitsPerChannel = m->mBitsPerChannel;
+    streamFormat.mBytesPerPacket = chan * buffer_bytes_per_channel;
+    streamFormat.mBytesPerFrame = chan * buffer_bytes_per_channel;
+
+    printf ("Stream format:\n");
+    printf (" Channels: %d\n", streamFormat.mChannelsPerFrame);
+    printf (" Sample rate: %f\n", streamFormat.mSampleRate);
+    printf (" Bits per channel: %d\n", streamFormat.mBitsPerChannel);
+    printf (" Bytes per frame: %d\n", streamFormat.mBytesPerFrame);
+
+    if (AudioUnitSetProperty (output_instance, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(streamFormat)))
+    {
+        fprintf (stderr, "Failed to set audio unit input property.\n");
+        return false;
+    }
+
+    if (AudioUnitSetProperty (output_instance, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, callback, sizeof (AURenderCallbackStruct)))
+    {
+        fprintf (stderr, "Unable to attach an IOProc to the selected audio unit.\n");
+        return false;
+    }
+
+    if (AudioOutputUnitStart (output_instance))
+    {
+        fprintf ("Unable to start audio unit.\n");
+        return false;
+    }
+
+    return true;
+}
+
+void close_audio (void)
+{
+    AudioOutputUnitStop (output_instance);
+}
+
+/*
+At this point you should have full playback with callback to your callback and
+shutdown. Now to implement volume control...
+*/
+
+/* value is 0..100, the actual applied volume is based on a natual decibel scale. */
+void set_volume (int value)
+{
+    float factor = (value == 0) ? 0.0 : powf (10, (float) VOLUME_RANGE * (value - 100) / 100 / 20);
+
+    /* lots of pain concerning controlling application volume can be avoided
+     * with this one neat trick... */
+    AudioUnitSetParameter (output_instance, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, factor, 0);
+}
+
+/*
+Two things here:
+
+Apple says you should set up an AUGraph for doing something as simple as controlling output volume. I say that is unnecessary. There is a lot of misinformation here as well, that kHALOutputParam_Volume sets the system volume; it doesn't. It sets the individual output volume on the sound server, coreaudiod.
+
+The reason for the powf () and the scary math is to give a logarithmic scale for tapering the volume down, similar to what would be observed in an actual stereo system. If you don't want this, just do float factor = value / 100.0.
+
+Now to handle pausing...
+*/
+
+void pause_audio (bool paused)
+{
+    if (paused)
+        AudioOutputUnitStop (output_instance);
+    else
+    {
+        if (AudioOutputUnitStart (output_instance))
+        {
+            fprintf (stderr, "Unable to restart audio unit after pausing.\n");
+            close_audio ();
+        }
+    }
+}
+
+} /* namespace coreaudio_example */
+
+/*
+That is all there is to the actual CoreAudio lowlevel operations. The rest is
+up to the reader. In the callback function you just copy your buffer data to
+ioData->mBuffers[0].mData. The amount of data requested is at
+ioData->mBuffers[0].mDataByteSize.
+
+Hopefully this documentation is useful to somebody else, I mostly wrote it for
+my own reference. Usually I get to just deal with FMOD which is much more
+pleasant (ha ha).
+*/