bcg729DecoderChannelContextStruct *initBcg729DecoderChannel()
{
/* create the context structure */
bcg729DecoderChannelContextStruct *decoderChannelContext = malloc(sizeof(bcg729DecoderChannelContextStruct));
/* intialise statics buffers and variables */
memcpy(decoderChannelContext->previousqLSP, previousqLSPInitialValues, NB_LSP_COEFF*sizeof(word16_t)); /* initialise the previousqLSP buffer */
memset(decoderChannelContext->excitationVector, 0, L_PAST_EXCITATION*sizeof(word16_t)); /* initialise the part of the excitationVector containing the past excitation */
decoderChannelContext->boundedAdaptativeCodebookGain = BOUNDED_PITCH_GAIN_MIN;
decoderChannelContext->pseudoRandomSeed = 21845; /* initialise pseudo Random seed according to spec 4.4.4 */
decoderChannelContext->adaptativeCodebookGain = 0; /* gains are initialised at 0 */
decoderChannelContext->fixedCodebookGain = 0;
memset(decoderChannelContext->reconstructedSpeech, 0, NB_LSP_COEFF*sizeof(word16_t)); /* initialise to zero all the values used from previous frame to get the current frame reconstructed speech */
/* initialisation of the differents blocs which need to be initialised */
initDecodeLSP(decoderChannelContext);
initDecodeAdaptativeCodeVector(decoderChannelContext);
initDecodeGains(decoderChannelContext);
initPostFilter(decoderChannelContext);
initPostProcessing(decoderChannelContext);
return decoderChannelContext;
}
status_t AudioALSAPlaybackHandlerNormal::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
AudioAutoTimeoutLock _l(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
// debug pcm dump
OpenPCMDump(LOG_TAG);
// acquire pmic clk
mHardwareResourceManager->EnableAudBufClk(true);
HpImpeDanceDetect();
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmI2S0Dl1Playback);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmI2S0Dl1Playback);
ALOGD("AudioALSAPlaybackHandlerNormal::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
pcm_params_free(params);
// HW attribute config // TODO(Harvey): query this
#ifdef PLAYBACK_USE_24BITS_ONLY
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_8_24_BIT;
#else
mStreamAttributeTarget.audio_format = (mStreamAttributeSource->audio_format == AUDIO_FORMAT_PCM_32_BIT) ? AUDIO_FORMAT_PCM_8_24_BIT : AUDIO_FORMAT_PCM_16_BIT;
#endif
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = ChooseTargetSampleRate(mStreamAttributeSource->sample_rate,mStreamAttributeSource->output_devices);
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 2;
mConfig.period_size = (mStreamAttributeTarget.buffer_size / (mConfig.channels * mConfig.period_count)) / ((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
// post processing
initPostProcessing();
#if defined(MTK_SPEAKER_MONITOR_SUPPORT)
unsigned int fc, bw;
int th;
if (mAudioFilterManagerHandler)
{
AudioALSASpeakerMonitor::getInstance()->GetFilterParam(&fc, &bw, &th);
ALOGD("%s(), fc %d bw %d, th %d", __FUNCTION__, fc, bw, th);
mAudioFilterManagerHandler->setSpkFilterParam(fc, bw, th);
}
#endif
// SRC
initBliSrc();
// bit conversion
initBitConverter();
initDataPending();
// disable lowjitter mode
SetLowJitterMode(true, mStreamAttributeTarget.sample_rate);
// open pcm driver
openPcmDriver(pcmindex);
// open codec driver
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
//============Voice UI&Unlock REFERECE=============
AudioVUnlockDL *VUnlockhdl = AudioVUnlockDL::getInstance();
if (VUnlockhdl != NULL)
{
VUnlockhdl->SetInputStandBy(false);
VUnlockhdl-> GetSRCInputParameter(mStreamAttributeTarget.sample_rate, mStreamAttributeTarget.num_channels, mStreamAttributeTarget.audio_format);
VUnlockhdl->GetFirstDLTime();
}
//===========================================
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
status_t AudioALSAPlaybackHandlerOffload::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
ALOGD("%s(), mStreamAttributeSource: format = %d",__FUNCTION__, mStreamAttributeSource->audio_format);
AudioAutoTimeoutLock _l(*AudioALSADriverUtility::getInstance()->getStreamSramDramLock());
// debug pcm dump
OpenPCMDump(LOG_TAG);
// acquire pmic clk
mHardwareResourceManager->EnableAudBufClk(true);
//doug to check
//HpImpeDanceDetect(); //doug to check
#if 1
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmDl1Meida);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmDl1Meida);
ALOGD("AudioALSAPlaybackHandlerOffload::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
ALOGD("%s buffersizemax = %d", __FUNCTION__, mStreamAttributeTarget.buffer_size);
pcm_params_free(params);
#endif
//mStreamAttributeTarget.buffer_size = 32768;
//#ifdef PLAYBACK_USE_24BITS_ONLY
//mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_32_BIT;
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_8_24_BIT;
//mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_16_BIT;
//#else
// mStreamAttributeTarget.audio_format = (mStreamAttributeSource->audio_format == AUDIO_FORMAT_PCM_32_BIT) ? AUDIO_FORMAT_PCM_8_24_BIT : AUDIO_FORMAT_PCM_16_BIT;
//#endif
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_STEREO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = ChooseTargetSampleRate(mStreamAttributeSource->sample_rate);
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 2;
mConfig.period_size = (mStreamAttributeTarget.buffer_size / (mConfig.channels * mConfig.period_count)) / ((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
mComprConfig.codec = (struct snd_codec*)malloc(sizeof(struct snd_codec));
if(mComprConfig.codec == NULL)
ALOGE("%s(), allocate mComprConfig.codec fail");
mComprConfig.fragments = 1024;
mComprConfig.fragment_size = 8192;
//mComprConfig.fragment_size = mStreamAttributeTarget.buffer_size;
mComprConfig.codec->sample_rate = mStreamAttributeTarget.sample_rate;
mComprConfig.codec->reserved[0] = mConfig.period_size;
mComprConfig.codec->reserved[1] = mComprConfig.fragments*mComprConfig.fragment_size;
if(mConfig.format == PCM_FORMAT_S16_LE)
mComprConfig.codec->format = SNDRV_PCM_FORMAT_S16_LE;
else
mComprConfig.codec->format = SNDRV_PCM_FORMAT_S32_LE;
mComprConfig.codec->id = SND_AUDIOCODEC_MP3;
mComprConfig.codec->ch_in = 2;
mComprConfig.codec->ch_out = 2;
//init decoder
mDecHandler = AudioDecHandlerCreate();
if(mDecHandler == NULL)
{
ALOGE("+%s(), DecHandler create fail", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
if(!mDecHandler->InitAudioDecoder())
{
ALOGE("+%s(), Decoder IP init fail", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
// post processing
initPostProcessing();
// SRC
initBliSrc();
// bit conversion
initBitConverter();
initDataPending();
// disable lowjitter mode //doug to check
SetLowJitterMode(true, mStreamAttributeTarget.sample_rate);
openComprDriver(23);
if( compress_set_gapless_metadata(mComprStream, &offload_stream.offload_mdata) != 0)
ALOGE("%s(), compress_set_gapless_metadata() error= %s", __FUNCTION__, compress_get_error(mComprStream));
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
offload_stream.tmpBuffer = (void*)malloc(mComprConfig.fragment_size);
mWritebytes = mComprConfig.fragment_size;
list_init(&offload_stream.offload_cmd_list);
int ret = pthread_mutex_init(&offload_stream.offload_mutex, NULL);
if (ret != 0)
{
ALOGE("%s, Failed to initialize Mutex!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
ret = pthread_cond_init(&offload_stream.offload_cond, NULL);
if (ret != 0)
{
ALOGE("%s, Failed to initialize Cond!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
threadExit = false;
ret = pthread_create(&offload_stream.offload_pthread, NULL, &offload_threadloop, this);
if (ret != 0)
{
ALOGE("%s() create thread OffloadWrite fail!!", __FUNCTION__);
ASSERT(false);
return -ENOSYS;
}
usleep(1 * 1000);
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
status_t AudioALSAPlaybackHandlerSphDL::open()
{
ALOGD("+%s(), mDevice = 0x%x", __FUNCTION__, mStreamAttributeSource->output_devices);
// debug pcm dump
OpenPCMDump(LOG_TAG);
int pcmindex = AudioALSADeviceParser::getInstance()->GetPcmIndexByString(keypcmI2S0Dl1Playback);
int cardindex = AudioALSADeviceParser::getInstance()->GetCardIndexByString(keypcmI2S0Dl1Playback);
ALOGD("AudioALSAPlaybackHandlerSphDL::open() pcmindex = %d", pcmindex);
ListPcmDriver(cardindex, pcmindex);
struct pcm_params *params;
params = pcm_params_get(cardindex, pcmindex, PCM_OUT);
if (params == NULL)
{
ALOGD("Device does not exist.\n");
}
mStreamAttributeTarget.buffer_size = pcm_params_get_max(params, PCM_PARAM_BUFFER_BYTES);
ALOGD("buffersizemax = %d", mStreamAttributeTarget.buffer_size);
pcm_params_free(params);
// HW attribute config // TODO(Harvey): query this
mStreamAttributeTarget.audio_format = AUDIO_FORMAT_PCM_16_BIT;
mStreamAttributeTarget.audio_channel_mask = AUDIO_CHANNEL_IN_MONO;
mStreamAttributeTarget.num_channels = android_audio_legacy::AudioSystem::popCount(mStreamAttributeTarget.audio_channel_mask);
mStreamAttributeTarget.sample_rate = mStreamAttributeSource->sample_rate; // same as source stream
// HW pcm config
mConfig.channels = mStreamAttributeTarget.num_channels;
mConfig.rate = mStreamAttributeTarget.sample_rate;
// Buffer size: 1536(period_size) * 2(ch) * 4(byte) * 2(period_count) = 24 kb
mConfig.period_count = 4;
//mConfig.period_size = (0x4000/(mConfig.channels*mConfig.period_count))/((mStreamAttributeTarget.audio_format == AUDIO_FORMAT_PCM_16_BIT) ? 2 : 4);
mConfig.period_size = 512;
mConfig.format = transferAudioFormatToPcmFormat(mStreamAttributeTarget.audio_format);
mConfig.start_threshold = 0;
mConfig.stop_threshold = 0;
mConfig.silence_threshold = 0;
ALOGD("%s(), mConfig: channels = %d, rate = %d, period_size = %d, period_count = %d, format = %d",
__FUNCTION__, mConfig.channels, mConfig.rate, mConfig.period_size, mConfig.period_count, mConfig.format);
// post processing
initPostProcessing();
#if defined(MTK_SPEAKER_MONITOR_SUPPORT)
unsigned int fc, bw;
int th;
if (mAudioFilterManagerHandler)
{
AudioALSASpeakerMonitor::getInstance()->GetFilterParam(&fc, &bw, &th);
ALOGD("%s(), fc %d bw %d, th %d", __FUNCTION__, fc, bw, th);
mAudioFilterManagerHandler->setSpkFilterParam(fc, bw, th);
}
#endif
// SRC
initBliSrc();
// bit conversion
initBitConverter();
// open pcm driver
openPcmDriver(pcmindex);
//Echo reference path
if (mixer_ctl_set_enum_by_string(mixer_get_ctl_by_name(mMixer, "Audio_Dl1_MD_Echo_Ref_Switch"), "On"))
{
ALOGE("Error: Audio_Dl1_MD_Echo_Ref_Switch invalid value");
}
// open codec driver
// Don't startoutputDevice here, let speech driver to open.
mHardwareResourceManager->startOutputDevice(mStreamAttributeSource->output_devices, mStreamAttributeTarget.sample_rate);
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
void scene() {
foreach(Planet * planet, planets)
planet->init();
initCamAndLight();
initPostProcessing();
}
