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Initial audio interface classes

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This commit is contained in:
Jack Andersen
2016-01-28 13:53:51 -10:00
parent be37b22dd5
commit 43376ff416
8 changed files with 397 additions and 7 deletions
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163
lib/audiodev/ALSA.cpp Normal file
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#include "boo/audiodev/IAudioVoiceAllocator.hpp"
#include <alsa/asoundlib.h>
#include <LogVisor/LogVisor.hpp>
namespace boo
{
static LogVisor::LogModule Log("boo::ALSA");
struct ALSAAudioVoice : IAudioVoice
{
ChannelMap m_map;
IAudioVoiceCallback* m_cb;
snd_pcm_t* m_pcm = nullptr;
snd_async_handler_t* m_handler = nullptr;
snd_pcm_uframes_t m_bufSize;
snd_pcm_uframes_t m_periodSize;
const ChannelMap& channelMap() const {return m_map;}
static void Callback(snd_async_handler_t* handler)
{
ALSAAudioVoice* voice = static_cast<ALSAAudioVoice*>(snd_async_handler_get_callback_private(handler));
voice->m_cb->needsNextBuffer(voice, voice->m_periodSize);
}
ALSAAudioVoice(AudioChannelSet set, unsigned sampleRate, IAudioVoiceCallback* cb)
: m_cb(cb)
{
if (snd_pcm_open(&m_pcm, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_ASYNC) < 0)
{
Log.report(LogVisor::Error, "unable to allocate ALSA voice");
return;
}
unsigned chCount = ChannelCount(set);
int err;
while ((err = snd_pcm_set_params(m_pcm, SND_PCM_FORMAT_S16, SND_PCM_ACCESS_RW_INTERLEAVED,
chCount, sampleRate, 1, 100000)) < 0)
{
if (set == AudioChannelSet::Stereo)
break;
set = AudioChannelSet(int(set) - 1);
chCount = ChannelCount(set);
}
if (err < 0)
{
snd_pcm_close(m_pcm);
m_pcm = nullptr;
Log.report(LogVisor::Error, "unable to set ALSA voice params");
return;
}
snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(m_pcm);
if (!chmaps)
{
snd_pcm_close(m_pcm);
m_pcm = nullptr;
Log.report(LogVisor::Error, "unable to query ALSA voice chmaps");
return;
}
snd_pcm_chmap_t* foundChmap = nullptr;
for (snd_pcm_chmap_query_t** chmap = chmaps ; *chmap != nullptr ; ++chmap)
{
if ((*chmap)->map.channels == chCount)
{
snd_pcm_chmap_t* chm = &(*chmap)->map;
uint64_t chBits = 0;
for (int c=0 ; c<chm->channels ; ++c)
chBits |= 1 << chm->pos[c];
bool good = false;
switch (set)
{
case AudioChannelSet::Stereo:
if ((chBits & (1 << SND_CHMAP_FL)) != 0 &&
(chBits & (1 << SND_CHMAP_FR)) != 0)
good = true;
break;
case AudioChannelSet::Quad:
if ((chBits & (1 << SND_CHMAP_FL)) != 0 &&
(chBits & (1 << SND_CHMAP_FR)) != 0 &&
(chBits & (1 << SND_CHMAP_RL)) != 0 &&
(chBits & (1 << SND_CHMAP_RR)) != 0)
good = true;
break;
case AudioChannelSet::Surround51:
if ((chBits & (1 << SND_CHMAP_FL)) != 0 &&
(chBits & (1 << SND_CHMAP_FR)) != 0 &&
(chBits & (1 << SND_CHMAP_RL)) != 0 &&
(chBits & (1 << SND_CHMAP_RR)) != 0 &&
(chBits & (1 << SND_CHMAP_FC)) != 0 &&
(chBits & (1 << SND_CHMAP_LFE)) != 0)
good = true;
break;
case AudioChannelSet::Surround71:
if ((chBits & (1 << SND_CHMAP_FL)) != 0 &&
(chBits & (1 << SND_CHMAP_FR)) != 0 &&
(chBits & (1 << SND_CHMAP_RL)) != 0 &&
(chBits & (1 << SND_CHMAP_RR)) != 0 &&
(chBits & (1 << SND_CHMAP_FC)) != 0 &&
(chBits & (1 << SND_CHMAP_LFE)) != 0 &&
(chBits & (1 << SND_CHMAP_SL)) != 0 &&
(chBits & (1 << SND_CHMAP_SR)) != 0)
good = true;
break;
}
if (good)
{
foundChmap = chm;
break;
}
}
}
if (!foundChmap)
{
snd_pcm_close(m_pcm);
m_pcm = nullptr;
snd_pcm_free_chmaps(chmaps);
Log.report(LogVisor::Error, "unable to find matching ALSA voice chmap");
return;
}
m_map.m_channelCount = chCount;
for (int c=0 ; c<foundChmap->channels ; ++c)
m_map.m_channels[c] = AudioChannel(foundChmap->pos[c] - 3);
snd_pcm_set_chmap(m_pcm, foundChmap);
snd_pcm_free_chmaps(chmaps);
snd_async_add_pcm_handler(&m_handler, m_pcm, snd_async_callback_t(Callback), this);
snd_pcm_get_params(m_pcm, &m_bufSize, &m_periodSize);
}
~ALSAAudioVoice()
{
if (m_handler)
snd_async_del_handler(m_handler);
if (m_pcm)
snd_pcm_close(m_pcm);
}
void bufferSampleData(const int16_t* data, size_t frames)
{
if (m_pcm)
snd_pcm_writei(m_pcm, data, frames);
}
};
struct ALSAAudioVoiceAllocator : IAudioVoiceAllocator
{
std::unique_ptr<IAudioVoice> allocateNewVoice(AudioChannelSet layoutOut,
unsigned sampleRate,
IAudioVoiceCallback* cb)
{
ALSAAudioVoice* newVoice = new ALSAAudioVoice(layoutOut, sampleRate, cb);
std::unique_ptr<IAudioVoice> ret(newVoice);
if (!newVoice->m_pcm)
return {};
return ret;
}
};
}

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lib/audiodev/AQS.cpp Normal file
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73
lib/audiodev/AudioMatrix.cpp Normal file
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#include "boo/audiodev/AudioMatrix.hpp"
#include <string.h>
namespace boo
{
void AudioMatrixMono::setDefaultMatrixCoefficients()
{
memset(m_coefs, 0, sizeof(m_coefs));
switch (m_setOut)
{
case AudioChannelSet::Mono:
case AudioChannelSet::Stereo:
case AudioChannelSet::Quad:
m_coefs[AudioChannel::FrontLeft] = 1.0;
m_coefs[AudioChannel::FrontRight] = 1.0;
break;
case AudioChannelSet::Surround51:
case AudioChannelSet::Surround71:
m_coefs[AudioChannel::FrontCenter] = 1.0;
break;
}
}
void AudioMatrixMono::bufferMonoSampleData(IAudioVoice& voice, const int16_t* data, size_t samples)
{
const ChannelMap& chmap = voice.channelMap();
m_interleaveBuf.clear();
m_interleaveBuf.reserve(samples * chmap.m_channelCount);
for (size_t s=0 ; s<samples ; ++s, ++data)
for (int c=0 ; c<chmap.m_channelCount ; ++c)
m_interleaveBuf.push_back(data[0] * m_coefs[chmap.m_channels[c]]);
voice.bufferSampleData(m_interleaveBuf.data(), samples);
}
void AudioMatrixStereo::setDefaultMatrixCoefficients()
{
memset(m_coefs, 0, sizeof(m_coefs));
switch (m_setOut)
{
case AudioChannelSet::Mono:
m_coefs[AudioChannel::FrontLeft][0] = 0.5;
m_coefs[AudioChannel::FrontLeft][1] = 0.5;
m_coefs[AudioChannel::FrontRight][0] = 0.5;
m_coefs[AudioChannel::FrontRight][1] = 0.5;
break;
case AudioChannelSet::Stereo:
case AudioChannelSet::Quad:
m_coefs[AudioChannel::FrontLeft][0] = 1.0;
m_coefs[AudioChannel::FrontRight][1] = 1.0;
break;
case AudioChannelSet::Surround51:
case AudioChannelSet::Surround71:
m_coefs[AudioChannel::FrontLeft][0] = 1.0;
m_coefs[AudioChannel::FrontRight][1] = 1.0;
break;
}
}
void AudioMatrixStereo::bufferStereoSampleData(IAudioVoice& voice, const int16_t* data, size_t frames)
{
const ChannelMap& chmap = voice.channelMap();
m_interleaveBuf.clear();
m_interleaveBuf.reserve(frames * chmap.m_channelCount);
for (size_t f=0 ; f<frames ; ++f, data += 2)
for (int c=0 ; c<chmap.m_channelCount ; ++c)
m_interleaveBuf.push_back(data[0] * m_coefs[chmap.m_channels[c][0]] +
data[1] * m_coefs[chmap.m_channels[c][1]]);
voice.bufferSampleData(m_interleaveBuf.data(), frames);
}
}

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lib/audiodev/XAudio2.cpp Normal file
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