/*****************************************************************************
                    The Dark Mod GPL Source Code
 
 This file is part of the The Dark Mod Source Code, originally based 
 on the Doom 3 GPL Source Code as published in 2011.
 
 The Dark Mod Source Code is free software: you can redistribute it 
 and/or modify it under the terms of the GNU General Public License as 
 published by the Free Software Foundation, either version 3 of the License, 
 or (at your option) any later version. For details, see LICENSE.TXT.
 
 Project: The Dark Mod (http://www.thedarkmod.com/)
 
 $Revision: 6529 $ (Revision of last commit) 
 $Date: 2015-08-14 17:47:10 +0000 (Fri, 14 Aug 2015) $ (Date of last commit)
 $Author: stevel $ (Author of last commit)
 
******************************************************************************/
#include "../../idlib/precompiled.h"
#include "../../sound/snd_local.h"
#include "../posix/posix_public.h"
#include "sound.h"

#include <dlfcn.h>

static idCVar s_alsa_pcm( "s_alsa_pcm", "default", CVAR_SYSTEM | CVAR_ARCHIVE, "which alsa pcm device to use. default, hwplug, hw.. see alsa docs" );
static idCVar s_alsa_lib( "s_alsa_lib", "libasound.so.2", CVAR_SYSTEM | CVAR_ARCHIVE, "alsa client sound library" );

/**
This idCVar configures how many extra frames of audio data it sends when the alsa driver 
experiences an underrun. This is more of a work around then a fix, as the root cause of 
the issue is likely elsewhere. -- NagaHuntress #4191
 */
static idCVar s_alsa_underrun_extrafill( "s_alsa_underrun_extrafill", "1024", CVAR_SYSTEM | CVAR_ARCHIVE | CVAR_NOCHEAT | CVAR_INTEGER, 
	"If an underrun error occurs while outputing ausio, it will retry and refill the audio stream with extra data to try and clear the underrun condition. This specifies the number of extra frames.", 
	0, 8192 );

/*
===============
idAudioHardwareALSA::DLOpen
===============
*/
bool idAudioHardwareALSA::DLOpen( void ) {
	const char *version;
	
	if ( m_handle ) {
		return true;
	}
	common->Printf( "dlopen(%s)\n", s_alsa_lib.GetString() );
	if ( !( m_handle = dlopen( s_alsa_lib.GetString(), RTLD_NOW | RTLD_GLOBAL ) ) ) {
		common->Printf( "dlopen(%s) failed: %s\n", s_alsa_lib.GetString(), dlerror() );
		return false;
	}
	// print the version if available
	id_snd_asoundlib_version = ( pfn_snd_asoundlib_version )dlsym( m_handle, "snd_asoundlib_version" );
	if ( !id_snd_asoundlib_version ) {
		common->Printf( "dlsym(\"snd_asoundlib_version\") failed: %s\n", dlerror() );
		common->Warning( "please consider upgrading alsa to a more recent version." );
	} else {
		version = id_snd_asoundlib_version();	
		common->Printf( "asoundlib version: %s\n", version );
	}
	// dlsym the symbols
	ALSA_DLSYM(snd_pcm_avail_update);
	ALSA_DLSYM(snd_pcm_close);
	ALSA_DLSYM(snd_pcm_hw_params);
	ALSA_DLSYM(snd_pcm_hw_params_any);
	ALSA_DLSYM(snd_pcm_hw_params_get_buffer_size);
	ALSA_DLSYM(snd_pcm_hw_params_set_access);
	ALSA_DLSYM(snd_pcm_hw_params_set_buffer_size_min);
	ALSA_DLSYM(snd_pcm_hw_params_set_channels);
	ALSA_DLSYM(snd_pcm_hw_params_set_format);
	ALSA_DLSYM(snd_pcm_hw_params_set_rate);
	ALSA_DLSYM(snd_pcm_hw_params_sizeof);
	ALSA_DLSYM(snd_pcm_open);
	ALSA_DLSYM(snd_pcm_prepare);
	ALSA_DLSYM(snd_pcm_state);
	ALSA_DLSYM(snd_pcm_writei);
	ALSA_DLSYM(snd_strerror);
	return true;
}

/*
===============
idAudioHardwareALSA::Release
===============
*/
void idAudioHardwareALSA::Release() {
	if ( m_pcm_handle ) {
		common->Printf( "close pcm\n" );
		id_snd_pcm_close( m_pcm_handle );
		m_pcm_handle = NULL;
	}
	if ( m_buffer ) {
		free( m_buffer );
		m_buffer = NULL;
	}
	if ( m_handle ) {
		common->Printf( "dlclose\n" );
		dlclose( m_handle );
		m_handle = NULL;
	}
}

/*
=================
idAudioHardwareALSA::InitFailed
=================	
*/	
void idAudioHardwareALSA::InitFailed() {
	Release();
	cvarSystem->SetCVarBool( "s_noSound", true );
	common->Warning( "sound subsystem disabled" );
	common->Printf( "--------------------------------------\n" );
}

/*
=====================
idAudioHardwareALSA::Initialize
=====================
*/
bool idAudioHardwareALSA::Initialize( void ) {
	int err;
	
	common->Printf( "------ Alsa Sound Initialization -----\n" );
	if ( !DLOpen() ) {
		InitFailed();
		return false;
	}
	if ( ( err = id_snd_pcm_open( &m_pcm_handle, s_alsa_pcm.GetString(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) < 0 ) {
		common->Printf( "snd_pcm_open SND_PCM_STREAM_PLAYBACK '%s' failed: %s\n", s_alsa_pcm.GetString(), id_snd_strerror( err ) );
		InitFailed();
		return false;
	}
	common->Printf( "opened Alsa PCM device %s for playback\n", s_alsa_pcm.GetString() );

	// set hardware parameters ----------------------------------------------------------------------

	// init hwparams with the full configuration space
	snd_pcm_hw_params_t *hwparams;
	// this one is a define
	id_snd_pcm_hw_params_alloca( &hwparams );
	if ( ( err = id_snd_pcm_hw_params_any( m_pcm_handle, hwparams ) ) < 0 ) {
		common->Printf( "cannot configure the PCM device: %s\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	if ( ( err = id_snd_pcm_hw_params_set_access( m_pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 ) {
		common->Printf( "SND_PCM_ACCESS_RW_INTERLEAVED failed: %s\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	if ( ( err = id_snd_pcm_hw_params_set_format( m_pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE ) ) < 0 ) {
		common->Printf( "SND_PCM_FORMAT_S16_LE failed: %s\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	// channels

	// sanity over number of speakers
	m_channels = idSoundSystemLocal::s_numberOfSpeakers.GetInteger();
	if ( m_channels != 6 && m_channels != 2 && m_channels != 8) {
		common->Warning( "invalid value for s_numberOfSpeakers. Use either 2, 6 or 8" );
		idSoundSystemLocal::s_numberOfSpeakers.SetInteger( 2 );
		m_channels = 2;
	}

	if ( ( err = id_snd_pcm_hw_params_set_channels( m_pcm_handle, hwparams, m_channels ) ) < 0 ) {
		common->Printf( "error setting %d channels: %s\n", m_channels, id_snd_strerror( err ) );
		if ( idSoundSystemLocal::s_numberOfSpeakers.GetInteger() != 2 ) {
			// fallback to stereo if that works
			m_channels = 2;
			if ( ( err = id_snd_pcm_hw_params_set_channels( m_pcm_handle, hwparams, m_channels ) ) < 0 ) {
				common->Printf( "fallback to stereo failed: %s\n", id_snd_strerror( err ) );
				InitFailed();
				return false;
			} else {
				common->Printf( "fallback to stereo\n" );
				idSoundSystemLocal::s_numberOfSpeakers.SetInteger( 2 );
			}
		} else {
			InitFailed();
			return false;
		}
	}

	// set sample rate (frequency)
	if ( ( err = id_snd_pcm_hw_params_set_rate( m_pcm_handle, hwparams, PRIMARYFREQ, 0 ) ) < 0 ) {
		common->Printf( "failed to set 44.1KHz rate: %s - try ( +set s_alsa_pcm plughw:0 ? )\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	// have enough space in the input buffer for our MIXBUFFER_SAMPLE feedings and async ticks
	snd_pcm_uframes_t frames;
	frames = MIXBUFFER_SAMPLES + MIXBUFFER_SAMPLES / 3;
	if ( ( err = id_snd_pcm_hw_params_set_buffer_size_min( m_pcm_handle, hwparams, &frames ) ) < 0 ) {
		common->Printf( "buffer size select failed: %s\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	// apply parameters
	if ( ( err = id_snd_pcm_hw_params( m_pcm_handle, hwparams ) ) < 0 ) {
		common->Printf( "snd_pcm_hw_params failed: %s\n", id_snd_strerror( err ) );
		InitFailed();
		return false;
	}

	// check the buffer size
	if ( ( err = id_snd_pcm_hw_params_get_buffer_size( hwparams, &frames ) ) < 0 ) {
		common->Printf( "snd_pcm_hw_params_get_buffer_size failed: %s\n", id_snd_strerror( err ) );
	} else {
		common->Printf( "device buffer size: %lu frames ( %lu bytes )\n", ( long unsigned int )frames, frames * m_channels * 2 );
	}

	// TODO: can use swparams to setup the device so it doesn't underrun but rather loops over
	// snd_pcm_sw_params_set_stop_threshold
	// To get alsa to just loop on underruns. set the swparam stop_threshold to equal buffer size. The sound buffer will just loop and never throw an xrun.

	// allocate the final mix buffer
	m_buffer_size = MIXBUFFER_SAMPLES * m_channels * 2;
	m_buffer = malloc( m_buffer_size );
	common->Printf( "allocated a mix buffer of %d bytes\n", m_buffer_size );

#ifdef _DEBUG
	// verbose the state
	snd_pcm_state_t curstate = id_snd_pcm_state( m_pcm_handle );
	assert( curstate == SND_PCM_STATE_PREPARED );
#endif
	
	common->Printf( "--------------------------------------\n" );
	return true;
}

/*
===============
idAudioHardwareALSA::~idAudioHardwareALSA
===============
*/
idAudioHardwareALSA::~idAudioHardwareALSA() {
	common->Printf( "----------- Alsa Shutdown ------------\n" );
	Release();
	common->Printf( "--------------------------------------\n" );
}

/*
=================
idAudioHardwareALSA::GetMixBufferSize
=================
*/	
int idAudioHardwareALSA::GetMixBufferSize() {
	return m_buffer_size;
}

/*
=================
idAudioHardwareALSA::GetMixBuffer
=================
*/	
short* idAudioHardwareALSA::GetMixBuffer() {
	return (short *)m_buffer;
}

/*
===============
idAudioHardwareALSA::Flush
===============
*/
bool idAudioHardwareALSA::Flush( void ) {
	int ret;
	snd_pcm_state_t state;
	state = id_snd_pcm_state( m_pcm_handle );
	if ( state != SND_PCM_STATE_RUNNING && state != SND_PCM_STATE_PREPARED ) {
		if ( ( ret = id_snd_pcm_prepare( m_pcm_handle ) ) < 0 ) {
			Sys_Printf( "failed to recover from SND_PCM_STATE_XRUN: %s\n", id_snd_strerror( ret ) );
			cvarSystem->SetCVarBool( "s_noSound", true );
			return false;
		}
		Sys_Printf( "preparing audio device for output\n" );
	}
	Write( true );
	return true;
}

/*
===============
idAudioHardwareALSA::Write
rely on m_freeWriteChunks which has been set in Flush() before engine did the mixing for this MIXBUFFER_SAMPLE
===============
*/
void idAudioHardwareALSA::Write( bool flushing ) {
	if ( !flushing && m_remainingFrames ) {
		// if we write after a new mixing loop, we should have m_writeChunk == 0
		// otherwise that last remaining chunk that was never flushed out to the audio device has just been overwritten
		Sys_Printf( "idAudioHardwareALSA::Write: %d frames overflowed and dropped\n", m_remainingFrames );
	}
	if ( !flushing ) {
		// if running after the mix loop, then we have a full buffer to write out
		m_remainingFrames = MIXBUFFER_SAMPLES;
	}
	if ( m_remainingFrames == 0 ) {
		return;
	}
	// write the max frames you can in one shot - we need to write it all out in Flush() calls before the next Write() happens
	size_t pos = (size_t)m_buffer + ( MIXBUFFER_SAMPLES - m_remainingFrames ) * m_channels * 2;
	snd_pcm_sframes_t frames = id_snd_pcm_writei( m_pcm_handle, (void*)pos, m_remainingFrames );
	/*
	  A kludge to handle buffer underruns (reported as broken pipes). 
	  To the user this condition manifests as popping or crackling audio.
	  This will re-ready the sound out and send the pending data twice.
	  The root cause of this error is likely to lie else where in the audio 
	  pipeline, but this work around plugs the problem for now. -- NagaHuntress #4191
	*/
	if ( frames == -EPIPE ) {
		snd_pcm_sframes_t nextframes = s_alsa_underrun_extrafill.GetInteger();
		nextframes = (nextframes < 0) ? 0 : nextframes;
		nextframes = (m_remainingFrames > nextframes) ? nextframes : m_remainingFrames;
		Sys_Printf( "snd_pcm_writei() reports broken pipe (underrun) while sending %u frames. Retrying but also sending %u duplicate frames first. Try increasing 's_alsa_underrun_extrafill' if this persists.\n", (unsigned int)m_remainingFrames, (unsigned int)nextframes);
		id_snd_pcm_prepare( m_pcm_handle );
		if(nextframes) {
			frames = id_snd_pcm_writei( m_pcm_handle, (void*)pos, nextframes );
		}
		frames = id_snd_pcm_writei( m_pcm_handle, (void*)pos, m_remainingFrames );
	}
	if ( frames < 0 ) {
		if ( frames != -EAGAIN ) {
			Sys_Printf( "snd_pcm_writei %d frames failed: %s\n", m_remainingFrames, id_snd_strerror( frames ) );
		}
		return;
	}
	m_remainingFrames -= frames;
}
