/****************************************************************************
DESCRIPTION
Disconnect Sync audio
*/
void CsrSbcEncoderPluginDisconnect( void )
{
uint16 i;
if (!SBC)
Panic() ;
StreamDisconnect(0, StreamKalimbaSink(0));
KalimbaPowerOff() ;
for (i = 0; i < MAX_AUDIO_SINKS; i++)
{
if (SBC->media_sink[i] != 0)
{
/* Disconnect the Kalimba source from the media sink */
StreamDisconnect(StreamKalimbaSource(i+2), SBC->media_sink[i]);
PRINT(("Audio Plugin: Disconnect media i:%d sink:0x%x\n",i,(uint16)SBC->media_sink[i]));
/* clear the audio sink */
SBC->media_sink[i] = 0;
}
}
free (SBC);
SBC = NULL ;
}
/****************************************************************************
DESCRIPTION
Stop prompt where DSP has not been loaded by the plugin, e.g. (adpcm or pcm)
Prompt is either mixing in an existing DSP app or not using the DSP.
*/
static void CsrVoicePresencesPluginStopPhraseMixable ( void )
{
Sink lSink=NULL;
Task taskdata = NULL;
/* Check for DSP mixing */
if(GetCurrentDspStatus())
{
lSink = StreamKalimbaSink(PRESENT_DSP_PORT);
/* reset the volume levels of the dsp plugin */
if(GetAudioPlugin())
MessageSend( GetAudioPlugin(), AUDIO_PLUGIN_RESET_VOLUME_MSG, 0 ) ;
}
else /* Must be ADPCM not mixing */
{
switch(koovox_phrase_data->features.audio_output_type)
{
/* is the I2S required? */
case OUTPUT_INTERFACE_TYPE_I2S:
{
CsrI2SAudioOutputDisconnect( koovox_phrase_data->features.stereo);
}
break;
/* spdif output? */
case OUTPUT_INTERFACE_TYPE_SPDIF:
{
Sink rSink = NULL;
/* obtain source to SPDIF hardware and disconnect it */
lSink = StreamAudioSink(AUDIO_HARDWARE_SPDIF, AUDIO_INSTANCE_0, SPDIF_CHANNEL_A );
rSink = StreamAudioSink(AUDIO_HARDWARE_SPDIF, AUDIO_INSTANCE_0, SPDIF_CHANNEL_B );
StreamDisconnect(0, lSink);
StreamDisconnect(0, rSink);
SinkClose(lSink);
SinkClose(rSink);
}
break;
/* use built in codec */
default:
{
lSink = StreamAudioSink(AUDIO_HARDWARE_CODEC,AUDIO_INSTANCE_0, (koovox_phrase_data->features.stereo ? AUDIO_CHANNEL_A_AND_B : AUDIO_CHANNEL_A));
/* Disconnect PCM source/sink */
StreamDisconnect(StreamSourceFromSink(lSink), lSink);
}
}
}
/* close sink and cancel any messages if valid */
if(lSink)
{
/* Cancel all the messages relating to VP that have been sent */
taskdata = MessageSinkTask(lSink, NULL);
SinkClose(lSink);
}
PRINT(("PRESENT: SinkTask now NULL was %x\n",(uint16)taskdata));
MessageCancelAll((TaskData*) &csr_voice_presences_plugin, MESSAGE_STREAM_DISCONNECT);
}
void CsrSubwooferPluginDisconnect(void)
{
/* Disconnect the appropriate input to Kalimba */
if (plugin_data->input == SUBWOOFER_INPUT_ADC)
{
PRINT(("[SW_PLUGIN : Disconnect ADC to Kalimba\n"));
StreamDisconnect(plugin_data->audio_source, StreamKalimbaSink(DSP_INPUT_PORT_ADC));
}
else if (plugin_data->input == SUBWOOFER_INPUT_ESCO)
{
PRINT(("[SW_PLUGIN] Disconnect eSCO to Kalimba\n"));
StreamDisconnect(plugin_data->audio_source, StreamKalimbaSink(DSP_INPUT_PORT_ESCO));
StreamConnectDispose(plugin_data->audio_source);
}
else if (plugin_data->input == SUBWOOFER_INPUT_L2CAP)
{
PRINT(("[SW_PLUGIN] Disconnect L2CAP to Kalimba\n"));
StreamDisconnect(plugin_data->audio_source, StreamKalimbaSink(DSP_INPUT_PORT_L2CAP));
StreamConnectDispose(plugin_data->audio_source);
}
/* Disconnect the appropriate output from Kalimba */
if (plugin_data->output == SUBWOOFER_OUTPUT_DAC)
{
PRINT(("[SW_PLUGIN] Disconnect Kalimba to DAC\n"));
StreamDisconnect(StreamKalimbaSource(DSP_OUTPUT_PORT_DAC), plugin_data->codec_sink);
}
else if (plugin_data->output == SUBWOOFER_OUTPUT_I2S)
{
PRINT(("[SW_PLUGIN] Disconnect Kalimba to I2S\n"));
CsrI2SAudioOutputDisconnect(TRUE);
}
/* dispose of any remaining messages in the queue */
(void) MessageCancelAll( (TaskData*)&csr_subwoofer_plugin, MESSAGE_FROM_KALIMBA);
MessageKalimbaTask( NULL );
/* Everything has been disconnected so power down Kalimba */
KalimbaPowerOff();
/* reset flags */
SetAudioInUse(FALSE);
SetCurrentDspStatus(DSP_NOT_LOADED);
/* Free the memory allocated to use for the plugin as it's no longer valid */
free(plugin_data);
plugin_data = NULL;
}
/****************************************************************************
DESCRIPTION
Stop a tone from currently playing
*/
void CsrA2dpDecoderPluginStopTone ( void )
{
if (!DECODER)
Panic() ;
StreamDisconnect ( 0 , StreamKalimbaSink(3) ) ;
}
/****************************************************************************
DESCRIPTION
Disconnect Sync audio
*/
void CsrA2dpDecoderPluginDisconnect( void )
{
if (!DECODER)
Panic() ;
/*disconnect the pcm streams*/
StreamDisconnect(StreamPcmSource(0), StreamPcmSink(0));
StreamDisconnect(StreamPcmSource(1), StreamPcmSink(1));
/* For sinks disconnect the source in case its currently being disposed. */
StreamDisconnect(StreamSourceFromSink(DECODER->media_sink ), 0);
StreamConnectDispose (StreamSourceFromSink(DECODER->media_sink)) ;
KalimbaPowerOff() ;
free (DECODER);
DECODER = NULL ;
}
void CsrSubwooferPluginConnect(Sink audio_sink, Task codec_task, Task app_task, subwooferPluginConnectParams * params)
{
FILE_INDEX index;
/* Update the current DSP status */
SetCurrentDspStatus(DSP_LOADING);
/* Give Kalimba the plugin task so it knows where to send messages */
(void) MessageCancelAll( (TaskData*)&csr_subwoofer_plugin, MESSAGE_FROM_KALIMBA);
MessageKalimbaTask( (TaskData*)&csr_subwoofer_plugin );
/* Load the Subwofoer DSP application - Panic if it could not be loaded */
index = PanicFalse( FileFind(FILE_ROOT, kal, sizeof(kal) - 1) );
PanicFalse( KalimbaLoad(index) );
/* update current DSP status */
SetCurrentDspStatus(DSP_LOADED_IDLE);
/* Allocate memory to store the plugin data */
plugin_data = (subwooferPluginData*)PanicUnlessMalloc(sizeof(subwooferPluginData));
/* Initialise the plugin data based on parameters supplied by the application */
plugin_data->audio_source = StreamSourceFromSink(audio_sink);
plugin_data->swat_system_volume_db = params->swat_system_volume_db;
plugin_data->swat_trim_gain_db = params->swat_trim_gain_db;
plugin_data->adc_volume_index = params->adc_volume;
plugin_data->input = params->input;
plugin_data->output = params->output;
plugin_data->sample_rate = params->sample_rate;
plugin_data->adc_sample_rate = params->adc_sample_rate;
plugin_data->codec_task = codec_task;
plugin_data->app_task = app_task;
plugin_data->dsp_set_sample_rate = 0; /* set later in response to AUDIO_PLUGIN_SET_MODE_MSG */
/* Zero the codecs output gain */
CodecSetOutputGainNow(plugin_data->codec_task, 0, left_and_right_ch);
CodecSetInputGainNow(plugin_data->codec_task, 0, left_and_right_ch);
/* If using the ADC, set the ADC source */
if (plugin_data->input == SUBWOOFER_INPUT_ADC)
{
/* Get the ADC source */
plugin_data->audio_source = StreamAudioSource(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_0, AUDIO_CHANNEL_A);
}
/* Disconnect the source in case it's currently being disposed */
StreamDisconnect(StreamSourceFromSink(audio_sink), 0);
PRINT(("[SW_PLUGIN] : CsrSubwooferPluginConnect - complete\n"));
}
/****************************************************************************
NAME
hidConnDisconnected
DESCRIPTION
Called to put the specified connection into the disconnected state.
RETURNS
void
*/
void hidConnDisconnected(HID *hid, Sink sink)
{
hidConnection *conn = NULL;
/* Find connection structure */
if (((hid->connection[HID_CON_CONTROL].state == hidConConnected) || (hid->connection[HID_CON_CONTROL].state == hidConDisconnecting)) &&
(sink == hid->connection[HID_CON_CONTROL].con.sink))
conn = &hid->connection[HID_CON_CONTROL];
else if (((hid->connection[HID_CON_INTERRUPT].state == hidConConnected) || (hid->connection[HID_CON_INTERRUPT].state == hidConDisconnecting)) &&
(sink == hid->connection[HID_CON_INTERRUPT].con.sink))
conn = &hid->connection[HID_CON_INTERRUPT];
else
Panic();
/* Dispose of any messages left in the l2cap sources to ensure the buffers are destroyed */
StreamDisconnect(StreamSourceFromSink(sink), 0);
StreamConnectDispose(StreamSourceFromSink(sink));
/* Move connection into disconnected state */
conn->state = hidConDisconnected;
}
/****************************************************************************
DESCRIPTION
disconnect the subwoofer if currently connected
*/
void CsrA2dpDecoderPluginDisconnectSubwoofer(void)
{
DECODER_t * DECODER = CsrA2dpDecoderGetDecoderData();
/* ensure the decoder is loaded */
if(DECODER)
{
A2dpPluginConnectParams *codecData = (A2dpPluginConnectParams *) DECODER->params;
/* if the sub woofer is connected then disconnect it */
if(codecData->sub_connection_port != DSP_SUB_PORT_NOT_CONNECTED)
{
PRINT(("DECODER: CsrA2dpDecoderPluginDisconnect disconnect sub\n"));
/* disconnect kalimba port, this causes dsp app to switch clock source internally */
StreamDisconnect(StreamKalimbaSource(codecData->sub_connection_port), codecData->sub_sink );
/* update connected ports state */
codecData->sub_connection_port = DSP_SUB_PORT_NOT_CONNECTED;
}
}
}
/****************************************************************************
DESCRIPTION
Stop a prompt from currently playing
*/
void CsrVoicePresencesPluginStopPhrase ( void )
{
if(!koovox_phrase_data)
Panic();
PRINT(("PRESENT: Terminated\n"));
if(koovox_phrase_data->mixing)
{
/* If DSP already loaded and the prompt was mixed */
CsrVoicePresencesPluginStopPhraseMixable();
}
/* Make sure prompt source is disposed */
if(SourceIsValid(koovox_phrase_data->source))
StreamDisconnect(koovox_phrase_data->source, NULL);
/* Tidy up */
free(koovox_phrase_data);
koovox_phrase_data = NULL;
SetAudioBusy(NULL) ;
SetVpPlaying(FALSE);
}
void A2dpStartKalimbaStreaming(const A2DP *a2dp, uint16 media_sink)
{
FILE_INDEX index = FILE_NONE;
#ifdef BUILD_FOR_23FW
Transform t;
#else
Transform t, l_t, r_t;
Source l_src = StreamAudioSource(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_0, AUDIO_CHANNEL_A);
Source r_src = StreamAudioSource(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_1, AUDIO_CHANNEL_B);
Sink l_snk = StreamAudioSink(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_0, AUDIO_CHANNEL_A);
Sink r_snk = StreamAudioSink(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_1, AUDIO_CHANNEL_B);
#endif
/* load SBC codec */
index = FileFind(FILE_ROOT, sbc_encoder, sizeof(sbc_encoder)-1);
if (!KalimbaLoad(index))
/* codec load failure, Panic as the test isn't going to work now */
Panic();
/* Init and configure RTP */
t = TransformRtpSbcEncode(StreamKalimbaSource(0), (Sink)media_sink);
TransformConfigure(t, VM_TRANSFORM_RTP_SBC_ENCODE_PACKET_SIZE, 668);
TransformConfigure(t, VM_TRANSFORM_RTP_SBC_ENCODE_MANAGE_TIMING, FALSE);
(void)TransformStart(t);
/* Configure SBC encoding format */
(void)PanicFalse(KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_PARAMS, 0x00bd, 0, 0, 0));
(void)PanicFalse(KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_BITPOOL, 0x0030, 0, 0, 0));
#ifdef BUILD_FOR_23FW
StreamDisconnect(StreamPcmSource(0), StreamPcmSink(0));
StreamDisconnect(StreamPcmSource(1), StreamPcmSink(1));
/* set up ADCs */
(void)PcmClearAllRouting();
(void)PanicFalse(PcmRateAndRoute(0, PCM_NO_SYNC, 44100, 44100, VM_PCM_INTERNAL_A));
(void)PanicFalse(PcmRateAndRoute(1, 0, 44100, 44100, VM_PCM_INTERNAL_B));
(void)PanicFalse(StreamConnect(StreamPcmSource(0),StreamKalimbaSink(0)));
(void)PanicFalse(StreamConnect(StreamPcmSource(1),StreamKalimbaSink(1)));
#else
SourceClose(l_src);
SourceClose(r_src);
SinkClose(l_snk);
SinkClose(r_snk);
(void) SourceConfigure(l_src, STREAM_CODEC_INPUT_RATE, 44100);
(void) SourceConfigure(r_src, STREAM_CODEC_INPUT_RATE, 44100);
(void) SourceSynchronise(l_src, r_src);
/* set up ADCs */
l_t = StreamConnect(l_src, StreamKalimbaSink(0));
r_t = StreamConnect(r_src, StreamKalimbaSink(1));
(void)TransformStart(l_t);
(void)TransformStart(r_t);
#endif
/* Start decode */
(void) PanicFalse(KalimbaSendMessage(KALIMBA_MSG_GO,0,0,0,0));
}
/****************************************************************************
DESCRIPTION
configure the sub woofer and connect its audio if present
*/
void CsrA2dpDecoderPluginSetSubWoofer(AUDIO_SUB_TYPE_T sub_type, Sink sub_sink)
{
DECODER_t * DECODER = CsrA2dpDecoderGetDecoderData();
/* ensure the decoder is loaded */
if(DECODER)
{
A2dpPluginConnectParams *codecData = (A2dpPluginConnectParams *) DECODER->params;
/* update the decoder params with the new sub woofer status */
codecData->sub_woofer_type = sub_type;
codecData->sub_sink = sub_sink;
/* determine sub woofer type and connect (or disconnect if gone away) the sink
to the appropriate dsp port */
switch(sub_type)
{
/* SUB no longer available, disconnect port if still connected */
case AUDIO_SUB_WOOFER_NONE:
PRINT(("DECODER: Disconnect woofer\n" ));
/* if source is still connected then disconnect it */
if(codecData->sub_connection_port != DSP_SUB_PORT_NOT_CONNECTED)
{
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* mute the outputs whilst the subwoofer port is connected and
decoder restarted */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_ENABLE);
}
PRINT(("DECODER: Disconnect kalimba port %x\n", codecData->sub_connection_port));
StreamDisconnect(StreamKalimbaSource(codecData->sub_connection_port), 0);
/* update connected ports state */
codecData->sub_connection_port = DSP_SUB_PORT_NOT_CONNECTED;
/* set volume levels to desired level */
SubConnectedNowUnmuteVolume(DECODER);
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* unmute the outputs again */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_DISABLE);
}
}
break;
/* SUB using esco as its bt link, connect to dsp port 2 */
case AUDIO_SUB_WOOFER_ESCO:
PRINT(("DECODER: connect esco dsp port 2, sub_sink = %x\n", (uint16)sub_sink ));
/* ensure the correct ype of link is requested */
if(DECODER->sink_type != AUDIO_SINK_AV)
{
/* connect SUB output from dsp, source port 2, to esco sink */
if(codecData->sub_connection_port != DSP_ESCO_SUB_PORT)
{
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* mute the outputs whilst the subwoofer port is connected and
decoder restarted */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_ENABLE);
}
/* if not already connected */
if(codecData->sub_connection_port == DSP_L2CAP_SUB_PORT)
{
/* wrong port connected, disconnect it first */
StreamDisconnect(StreamKalimbaSource(codecData->sub_connection_port), 0);
PRINT(("DECODER: Disconnect kalimba port %x\n", codecData->sub_connection_port));
}
/* ensure sink is valid */
if(SinkIsValid(sub_sink))
{
/* connect esco (2) port */
if(StreamConnect(StreamKalimbaSource(DSP_ESCO_SUB_PORT),sub_sink))
{
/* update connected ports state */
codecData->sub_connection_port = DSP_ESCO_SUB_PORT;
PRINT(("DECODER: Connect kalimba port %x\n", codecData->sub_connection_port));
/* set volume levels to desired level */
SubConnectedNowUnmuteVolume(DECODER);
}
else
PRINT(("DECODER: connect esco dsp port 2 FAILED\n" ));
}
else
{
PRINT(("DECODER: connect esco dsp port 2, sub_sink = %x NOT VALID\n", (uint16)sub_sink ));
}
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* unmute the outputs again */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_DISABLE);
}
}
}
else
PRINT(("DECODER: ESCO - ****wrong link type****\n"));
break;
/* SUB using l2cap as its bt link, connect to dsp port 3 */
case AUDIO_SUB_WOOFER_L2CAP:
PRINT(("DECODER: connect l2cap dsp port 3, sub_sink = %x\n", (uint16)sub_sink ));
/* ensure the correct ype of link is requested */
if(DECODER->sink_type == AUDIO_SINK_AV)
{
/* connect SUB output from dsp, source port 2, to esco sink */
if(codecData->sub_connection_port != DSP_L2CAP_SUB_PORT)
{
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* mute the outputs whilst the subwoofer port is connected and
decoder restarted */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_ENABLE);
}
/* if not already connected */
if(codecData->sub_connection_port == DSP_ESCO_SUB_PORT)
{
/* wrong port connected, disconnect it first */
StreamDisconnect(StreamKalimbaSource(codecData->sub_connection_port), 0);
PRINT(("DECODER: Disconnect kalimba port %x\n", codecData->sub_connection_port));
}
/* ensure sink is valid */
if(SinkIsValid(sub_sink))
{
/* connect l2cap (3) port */
if(StreamConnect(StreamKalimbaSource(DSP_L2CAP_SUB_PORT),sub_sink))
{
/* update connected ports state */
codecData->sub_connection_port = DSP_L2CAP_SUB_PORT;
PRINT(("DECODER: Connect kalimba port %x\n", codecData->sub_connection_port));
/* set volume levels to desired level */
SubConnectedNowUnmuteVolume(DECODER);
}
else
PRINT(("DECODER: connect l2cap dsp port 3 FAILED\n" ));
}
else
{
PRINT(("DECODER: connect l2cap dsp port 3, sub_sink = %x NOT VALID\n", (uint16)sub_sink ));
}
/* determine if the subwoofer has connected after the decoder was loaded,
configured and unmuted */
if(codecData->sub_is_available == FALSE)
{
/* unmute the outputs again */
csrA2dpDecoderPluginOutputMute(multi_channel_group_all, AUDIO_MUTE_DISABLE);
}
}
}
else
PRINT(("DECODER: L2CAP - ****wrong link type****\n"));
break;
}
}
else
PRINT(("DECODER: CsrA2dpDecoderPluginSetSubWoofer ERROR NO DECODER\n" ));
}
/****************************************************************************
DESCRIPTION
This function connects a synchronous audio stream to the pcm subsystem
*/
void CsrSbcEncoderPluginConnect( Sink audio_sink , Task codec_task , uint16 volume , uint32 rate , bool stereo , AUDIO_MODE_T mode , const void * params )
{
/* DSP Application loading and Transform starting is handled by a call to A2dpAudioCodecEnable
in the application, so should not be done here. */
typedef struct
{
unsigned source_type:4;
unsigned reserved:4;
uint8 content_protection;
uint32 voice_rate;
unsigned bitpool:8;
unsigned format:8;
uint16 packet_size;
Sink media_sink_b;
} sbc_codec_data_type;
sbc_codec_data_type *sbc_codecData = (sbc_codec_data_type *) params;
if (!sbc_codecData)
Panic();
SBC = (SBC_t*)PanicUnlessMalloc (sizeof (SBC_t) ) ;
SBC->media_sink[0] = audio_sink ;
SBC->codec_task = codec_task ;
SBC->media_sink[1] = sbc_codecData->media_sink_b;
SBC->packet_size = sbc_codecData->packet_size;
StreamDisconnect(StreamKalimbaSource(2), 0);
/* Initialise the RTP SBC encoder */
SBC->t[0] = TransformRtpSbcEncode(StreamKalimbaSource(2), audio_sink);
/* Configure the RTP transform to generate the selected packet size */
TransformConfigure(SBC->t[0], VM_TRANSFORM_RTP_SBC_ENCODE_PACKET_SIZE, sbc_codecData->packet_size);
/* Transform should not manage timings. */
TransformConfigure(SBC->t[0], VM_TRANSFORM_RTP_SBC_ENCODE_MANAGE_TIMING, FALSE);
/* Start the transform */
(void) TransformStart(SBC->t[0]);
PRINT(("Audio Plugin: TransformStart sink:0x%x\n",(uint16)audio_sink));
if (SBC->media_sink[1])
{
/* There is a 2nd audio stream so initialise this transform */
StreamDisconnect(StreamKalimbaSource(3), 0);
/* Initialise the RTP SBC encoder */
SBC->t[1] = TransformRtpSbcEncode(StreamKalimbaSource(3), SBC->media_sink[1]);
/* Configure the RTP transform to generate the selected packet size */
TransformConfigure(SBC->t[1], VM_TRANSFORM_RTP_SBC_ENCODE_PACKET_SIZE, sbc_codecData->packet_size);
/* Transform should not manage timings. */
TransformConfigure(SBC->t[1], VM_TRANSFORM_RTP_SBC_ENCODE_MANAGE_TIMING, FALSE);
/* Start the transform */
(void) TransformStart(SBC->t[1]);
PRINT(("Audio Plugin: TransformStart sink:0x%x\n",(uint16)SBC->media_sink[1]));
}
/* Configure SBC encoding format */
if (!KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_PARAMS, sbc_codecData->format, 0, 0, 0))
/* If message fails to get through, abort */
Panic();
/* Pass bit pool value to DSP */
if (!KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_BITPOOL, sbc_codecData->bitpool, 0, 0, 0))
/* If message fails to get through, abort */
Panic();
/* disard any data sent by the SNK */
StreamConnectDispose(StreamSourceFromSink(audio_sink));
if (SBC->media_sink[1])
{
/* disard any data sent by the 2nd SNK */
StreamConnectDispose(StreamSourceFromSink(SBC->media_sink[1]));
}
/* select the source type */
if(sbc_codecData->source_type == SourceUsb)
{
PRINT(("Audio Plugin: SourceUsb\n"));
/* Select the source type */
PanicFalse(KalimbaSendMessage(KALIMBA_ENCODER_SELECT, 0x0001, 0, 0, 0));
}
else
{
PRINT(("Audio Plugin: SourceAnalog\n"));
/* For analogue input source */
StreamDisconnect(StreamPcmSource(0), StreamPcmSink(0));
StreamDisconnect(StreamPcmSource(1), StreamPcmSink(1));
(void)PcmClearAllRouting();
if (CodecGetCodecType(codec_task) == codec_wm8731)
{
PRINT(("Audio Plugin: codec_wm8731\n"));
/* configure slot 0 and 1 to be left and right channel
and synchronise the offsets for stereo playback */
(void)PcmRateAndRoute(0, PCM_NO_SYNC, (uint32) rate, (uint32) rate, VM_PCM_EXTERNAL_I2S);
(void)PcmRateAndRoute(1, 0, (uint32) rate, (uint32) rate, VM_PCM_EXTERNAL_I2S);
}
else
{
PRINT(("Audio Plugin: codec_internal\n"));
/* configure slot 0 and 1 to be left and right channel
and synchronise the offsets for stereo playback */
(void)PcmRateAndRoute(0, PCM_NO_SYNC, (uint32) rate, (uint32) rate, VM_PCM_INTERNAL_A);
(void)PcmRateAndRoute(1, 0, (uint32) rate, (uint32) rate, VM_PCM_INTERNAL_B);
}
/* plug Left ADC into port 0 */
(void)StreamConnect(StreamPcmSource(0),StreamKalimbaSink(0));
/* plug Right ADC into port 1 */
(void)StreamConnect(StreamPcmSource(1),StreamKalimbaSink(1));
/* Select the source type */
PanicFalse(KalimbaSendMessage(KALIMBA_ENCODER_SELECT, 0x0002, 0, 0, 0));
if(!KalimbaSendMessage(KALIMBA_MSG_GO,0,0,0,0))
{
PRINT(("SBC: Message KALIMBA_MSG_GO failed!\n"));
Panic();
}
}
/* Select the source type */
/*PanicFalse(KalimbaSendMessage(KALIMBA_SOURCE_SELECT, SOURCE_USB, 0, 0, 0)); */
/*CsrSbcEncoderUsbPluginSetVolume(volume) ;*/
/*(void) StreamConnect(PanicNull(StreamUsbEndPointSource(end_point_iso_in)), StreamKalimbaSink(0));
*/
/* Start decode */
/* if(!KalimbaSendMessage(KALIMBA_MSG_GO,0,0,0,0))
{
PRINT(("SBC: Message KALIMBA_MSG_GO failed!\n"));
Panic();
}
*/
}
/****************************************************************************
DESCRIPTION
Set the mode
*/
void CsrSbcEncoderPluginSetMode ( AUDIO_MODE_T mode , const void * params )
{
typedef struct
{
bool connect_sink;
Sink media_sink;
} sbc_mode_params;
sbc_mode_params *sbc_mode = (sbc_mode_params *) params;
uint16 i;
if (!SBC)
Panic() ;
PRINT(("Audio Plugin: Set Mode\n"));
if (sbc_mode->connect_sink)
{
/* A second audio sink has been connected */
for (i = 0; i < MAX_AUDIO_SINKS; i++)
{
if (SBC->media_sink[i] == 0)
{
/* store the audio sink */
SBC->media_sink[i] = sbc_mode->media_sink;
/* Initialise the RTP SBC encoder */
SBC->t[i] = TransformRtpSbcEncode(StreamKalimbaSource(i+2), SBC->media_sink[i]);
/* Configure the RTP transform to generate the selected packet size */
TransformConfigure(SBC->t[i], VM_TRANSFORM_RTP_SBC_ENCODE_PACKET_SIZE, SBC->packet_size);
/* Transform should not manage timings. */
TransformConfigure(SBC->t[i], VM_TRANSFORM_RTP_SBC_ENCODE_MANAGE_TIMING, FALSE);
/* Start the transform */
(void) TransformStart(SBC->t[i]);
PRINT(("Audio Plugin: Start new Transform i:%d sink:0x%x\n",i,(uint16)SBC->media_sink[i]));
}
}
}
else
{
/* An audio sink has been disconnected */
for (i = 0; i < MAX_AUDIO_SINKS; i++)
{
if (SBC->media_sink[i] == sbc_mode->media_sink)
{
/* Disconnect the Kalimba source from the media sink */
StreamDisconnect(StreamKalimbaSource(i+2), SBC->media_sink[i]);
PRINT(("Audio Plugin: Disconnect media i:%d sink:0x%x\n",i,(uint16)SBC->media_sink[i]));
/* clear the audio sink */
SBC->media_sink[i] = 0;
}
}
}
free(sbc_mode);
}
/****************************************************************************
DESCRIPTION
This function connects a synchronous audio stream to the pcm subsystem
*/
void CsrA2dpDecoderPluginConnect( A2dpPluginTaskdata *task, Sink audio_sink , Task codec_task , uint16 volume , uint32 rate , bool stereo , AUDIO_MODE_T mode , const void * params , Task app_task )
{
FILE_INDEX index = FILE_NONE;
char* kap_file = NULL ;
/* Only need to read the PS Key value once */
if (!pskey_read)
{
if (PsFullRetrieve(PSKEY_MAX_CLOCK_MISMATCH, &val_pskey_max_mismatch, sizeof(uint16)) == 0)
val_pskey_max_mismatch = 0;
pskey_read = TRUE;
}
switch ((A2DP_DECODER_PLUGIN_TYPE_T)task->a2dp_plugin_variant)
{
case SBC_DECODER:
kap_file = "sbc_decoder/sbc_decoder.kap";
break;
case MP3_DECODER:
kap_file = "mp3_decoder/mp3_decoder.kap";
break;
case AAC_DECODER:
kap_file = "aac_decoder/aac_decoder.kap";
break;
case FASTSTREAM_SINK:
kap_file = "faststream_sink/faststream_sink.kap";
break;
default:
Panic();
break;
}
/*ensure that the messages received are from the correct kap file*/
(void) MessageCancelAll( (TaskData*) task, MESSAGE_FROM_KALIMBA);
MessageKalimbaTask( (TaskData*) task );
index = FileFind(FILE_ROOT,(const char *) kap_file ,strlen(kap_file));
if (index == FILE_NONE)
Panic();
if (!KalimbaLoad(index))
Panic();
DECODER = (DECODER_t*)PanicUnlessMalloc (sizeof (DECODER_t) ) ;
DECODER->media_sink = audio_sink ;
DECODER->codec_task = codec_task ;
DECODER->volume = volume;
DECODER->mode = mode;
DECODER->stereo = stereo;
DECODER->params = (uint16) params;
DECODER->rate = rate;
DECODER->app_task = app_task;
if ((A2DP_DECODER_PLUGIN_TYPE_T)task->a2dp_plugin_variant == AAC_DECODER)
{
/* Workaround for AAC+ sources that negotiate sampling frequency at half the actual value */
if (rate < 32000)
DECODER->rate = rate * 2;
}
CodecSetOutputGainNow(DECODER->codec_task, 0, left_and_right_ch);
StreamDisconnect(StreamPcmSource(0), StreamPcmSink(0));
StreamDisconnect(StreamPcmSource(1), StreamPcmSink(1));
PanicFalse(PcmClearRouting(0));
PanicFalse(PcmClearRouting(1));
/* For sinks disconnect the source in case its currently being disposed. */
StreamDisconnect(StreamSourceFromSink(audio_sink), 0);
PRINT(("DECODER: CsrA2dpDecoderPluginConnect completed\n"));
}
/****************************************************************************
DESCRIPTION
Connect the encoded audio input and pcm audio output streams
*/
static void MusicConnectAudio (A2dpPluginTaskdata *task, bool stereo )
{
plugin_codec_data_type *codecData = (plugin_codec_data_type *) DECODER->params;
uint32 voice_rate = 0;
val_clock_mismatch = codecData->clock_mismatch;
if ((A2DP_DECODER_PLUGIN_TYPE_T)task->a2dp_plugin_variant == FASTSTREAM_SINK)
{
/*
FastStream does not use RTP.
L2CAP frames enter/leave via port 2
*/
/*
Initialise PCM.
Output stereo at 44k1Hz or 48kHz, input from left ADC at 16kHz.
*/
/* If no voice rate is set just make the ADC rate equal to 16kHz */
if (!codecData->voice_rate)
voice_rate = 16000;
else
voice_rate = codecData->voice_rate;
PRINT(("FASTSTREAM: rate=0x%lx voice_rate=0x%lx\n format=0x%x bitpool=0x%x",DECODER->rate,codecData->voice_rate,codecData->format,codecData->bitpool));
PanicFalse(PcmRateAndRoute(0, PCM_NO_SYNC, (uint32)DECODER->rate, (uint32) voice_rate, VM_PCM_INTERNAL_A));
/* is it mono playback? */
if ( !stereo )
{
PRINT(("DECODER: Mono\n"));
/* Connect Kalimba to PCM */
if (DECODER->rate)
{
StreamDisconnect(StreamSourceFromSink(DECODER->media_sink), 0);
PanicFalse(StreamConnect(StreamKalimbaSource(0),StreamPcmSink(0)));
PanicFalse(StreamConnect(StreamSourceFromSink(DECODER->media_sink),StreamKalimbaSink(2)));
}
}
else
{
PRINT(("DECODER: Stereo\n"));
PanicFalse(PcmRateAndRoute(1, 0, (uint32)DECODER->rate, (uint32) voice_rate, VM_PCM_INTERNAL_B));
/* Connect Kalimba to PCM */
if (DECODER->rate)
{
StreamDisconnect(StreamSourceFromSink(DECODER->media_sink), 0);
PanicFalse(StreamConnect(StreamKalimbaSource(0),StreamPcmSink(0)));
PanicFalse(StreamConnect(StreamKalimbaSource(1),StreamPcmSink(1)));
PanicFalse(StreamConnect(StreamSourceFromSink(DECODER->media_sink),StreamKalimbaSink(2)));
}
}
if (codecData->voice_rate)
{
StreamDisconnect(0, DECODER->media_sink);
/* configure parameters */
PanicFalse(KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_PARAMS, codecData->format, 0, 0, 0));
PanicFalse(KalimbaSendMessage(KALIMBA_MSG_SBCENC_SET_BITPOOL, codecData->bitpool, 0, 0, 0));
PanicFalse(StreamConnect(StreamPcmSource(0), StreamKalimbaSink(0)));
PanicFalse(StreamConnect(StreamKalimbaSource(2),DECODER->media_sink));
}
}
else /* Not FastStream CODEC */
{
uint8 content_protection = codecData->content_protection;
uint16 scms_enabled = 0;
Transform rtp_transform = 0;
switch ((A2DP_DECODER_PLUGIN_TYPE_T)task->a2dp_plugin_variant)
{
case SBC_DECODER:
rtp_transform = TransformRtpSbcDecode(StreamSourceFromSink(DECODER->media_sink) , StreamKalimbaSink(0));
break;
case MP3_DECODER:
rtp_transform = TransformRtpMp3Decode(StreamSourceFromSink(DECODER->media_sink) , StreamKalimbaSink(0));
break;
case AAC_DECODER:
rtp_transform = TransformRtpAacDecode(StreamSourceFromSink(DECODER->media_sink) , StreamKalimbaSink(0));
break;
default:
break;
}
/* Configure the content protection */
if (content_protection)
scms_enabled = 1;
TransformConfigure(rtp_transform, VM_TRANSFORM_RTP_SCMS_ENABLE, scms_enabled);
/*start the transform decode*/
(void)TransformStart( rtp_transform ) ;
/* is it mono playback? */
if ( !stereo )
{
PcmRateAndRoute(0, PCM_NO_SYNC, DECODER->rate, (uint32) 8000, VM_PCM_INTERNAL_A) ;
PRINT(("DECODER: Mono\n"));
/* plug port 0 into both DACs */
(void) PanicFalse(StreamConnect(StreamKalimbaSource(0),StreamPcmSink(0)));
}
else
{
PRINT(("DECODER: Stereo\n"));
PanicFalse(PcmRateAndRoute(0, PCM_NO_SYNC, DECODER->rate, (uint32) 8000, VM_PCM_INTERNAL_A));
PanicFalse(PcmRateAndRoute(1, 0, DECODER->rate, (uint32) 8000, VM_PCM_INTERNAL_B));
/* plug port 0 into Left DAC */
PanicFalse(StreamConnect(StreamKalimbaSource(0),StreamPcmSink(0)));
/* plug port 1 into Right DAC */
PanicFalse(StreamConnect(StreamKalimbaSource(1),StreamPcmSink(1)));
}
}
CsrA2dpDecoderPluginSetVolume(task,DECODER->volume) ;
/* The DSP must know the sample rate for tone mixing */
KalimbaSendMessage(MESSAGE_SET_SAMPLE_RATE, DECODER->rate, val_pskey_max_mismatch, val_clock_mismatch, 0);
PRINT(("DECODER: Send Go message to DSP now\n"));
if(!KalimbaSendMessage(KALIMBA_MSG_GO,0,0,0,0))
{
PRINT(("DECODER: Message KALIMBA_MSG_GO failed!\n"));
Panic();
}
}
/****************************************************************************
DESCRIPTION
This function connects APTX low delay audio
****************************************************************************/
void MusicConnectAptxLowLatency(A2dpPluginConnectParams *codecData, uint8 content_protection)
{
Sink speaker_snk_a = NULL;
Transform rtp_transform = 0;
DECODER_t * DECODER = CsrA2dpDecoderGetDecoderData();
/* when configured for using back channel low latency apps */
if(isCodecLowLatencyBackChannel())
{
/* if the back channel is required and the mic/line input is configured */
if (codecData->voice_rate)
{
/* Bidirectional channel uses faststream mono with the following settings */
/*
Configure the SBC format for the microphone data
16kHz, Mono, Blocks 16, Sub-bands 8, Loudness, Bitpool = 32
(data rate = 72kbps, packet size = 3*72 + 4 = 220 <= DM5).
*/
codecData->format = 0x31;
codecData->bitpool = 32;
PRINT(("DECODER: apt-X Low Latency Bidirectional rate=0x%lx voice_rate=0x%lx\n format=0x%x\n",DECODER->rate,codecData->voice_rate,codecData->format));
}
}
/* ensure the sample rate is valid */
if (DECODER->rate)
{
/* disconnect media sink as it might be disposed */
StreamDisconnect(StreamSourceFromSink(DECODER->media_sink), 0);
/* determine the output hardware type */
switch(DECODER->features.audio_output_type)
{
/* using the inbuilt dacs */
case OUTPUT_INTERFACE_TYPE_NONE:
case OUTPUT_INTERFACE_TYPE_DAC:
{
/* configure built-in audio hardware channel A */
speaker_snk_a = StreamAudioSink(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_0, AUDIO_CHANNEL_A);
PanicFalse(SinkConfigure(speaker_snk_a, STREAM_CODEC_OUTPUT_RATE, DECODER->rate));
/* if STEREO mode configured then connect the output channel B */
if(DECODER->features.stereo)
{
Sink speaker_snk_b = NULL;
PRINT(("DECODER: Stereo\n"));
/* connect channels B */
speaker_snk_b = StreamAudioSink(AUDIO_HARDWARE_CODEC, AUDIO_INSTANCE_0, AUDIO_CHANNEL_B);
/* configure channel to required rate */
PanicFalse(SinkConfigure(speaker_snk_b, STREAM_CODEC_OUTPUT_RATE, DECODER->rate));
/* synchronise both sinks for channels A & B */
PanicFalse(SinkSynchronise(speaker_snk_a, speaker_snk_b));
/* plug port 1 into Right DAC */
PanicFalse(StreamConnect(StreamKalimbaSource(AUDIO_OUT_FROM_DSP_LEFT),speaker_snk_a));
PanicFalse(StreamConnect(StreamKalimbaSource(AUDIO_OUT_FROM_DSP_RIGHT),speaker_snk_b));
}
/* mono operation, only connect left port */
else
{
/* plug port 0 into Left DAC */
PanicFalse(StreamConnect(StreamKalimbaSource(AUDIO_OUT_FROM_DSP_LEFT),speaker_snk_a));
}
}
break;
/* using the spdif digital output hardware */
case OUTPUT_INTERFACE_TYPE_SPDIF:
{
Sink speaker_snk_b = NULL;
/* configure spdif audio hardware channel 0 */
speaker_snk_a = StreamAudioSink(AUDIO_HARDWARE_SPDIF, AUDIO_INSTANCE_0, SPDIF_CHANNEL_A);
speaker_snk_b = StreamAudioSink(AUDIO_HARDWARE_SPDIF, AUDIO_INSTANCE_0, SPDIF_CHANNEL_B);
/* configure channel to required rate */
PanicFalse(SinkConfigure(speaker_snk_a, STREAM_SPDIF_OUTPUT_RATE, DECODER->rate));
PanicFalse(SinkConfigure(speaker_snk_b, STREAM_SPDIF_OUTPUT_RATE, DECODER->rate));
/* connect channels B */
/* synchronise both sinks for channels A & B */
PanicFalse(SinkSynchronise(speaker_snk_a, speaker_snk_b));
/* plug port 1 into Right DAC */
PanicFalse(StreamConnect(StreamKalimbaSource(AUDIO_OUT_FROM_DSP_LEFT),speaker_snk_a));
PanicFalse(StreamConnect(StreamKalimbaSource(AUDIO_OUT_FROM_DSP_RIGHT),speaker_snk_b));
PRINT(("DECODER: Stereo\n"));
}
break;
/* using the i2s digital output hardware */
case OUTPUT_INTERFACE_TYPE_I2S:
{
/* is a specified output frequency required? use resampling*/
if(CsrI2SMusicResamplingFrequency())
CsrI2SAudioOutputConnect(CsrI2SMusicResamplingFrequency(), DECODER->features.stereo, StreamKalimbaSource(AUDIO_OUT_FROM_DSP_LEFT), StreamKalimbaSource(AUDIO_OUT_FROM_DSP_RIGHT));
/* use the negotiated sample rate of the input, no resampling required */
else
CsrI2SAudioOutputConnect(DECODER->rate, DECODER->features.stereo, StreamKalimbaSource(AUDIO_OUT_FROM_DSP_LEFT), StreamKalimbaSource(AUDIO_OUT_FROM_DSP_RIGHT));
}
break;
}
/* if content protection is required feed the media sink through the rtp decoder transform */
if (content_protection)
{
rtp_transform = TransformRtpDecode(StreamSourceFromSink(DECODER->media_sink) , StreamKalimbaSink(LOW_LATENCY_CODEC_TO_DSP_PORT));
TransformConfigure(rtp_transform, VM_TRANSFORM_RTP_SCMS_ENABLE, content_protection);
/*start the transform decode*/
PRINT(("aptX: RTP Transform \n"));
(void)TransformStart( rtp_transform ) ;
}
/* connect the media sink to the dsp input port */
else
{
PanicFalse(StreamConnect(StreamSourceFromSink(DECODER->media_sink),StreamKalimbaSink(LOW_LATENCY_CODEC_TO_DSP_PORT)));
}
}
/* Send parameters that configure the SRA and buffer settings */
PRINT(("aptX LL params: initial level=%d target level=%d sra max rate=%d/10000 sra avg time=%d good working buffer level=%d \n",
codecData->aptx_sprint_params.target_codec_level,codecData->aptx_sprint_params.initial_codec_level,
codecData->aptx_sprint_params.sra_max_rate,codecData->aptx_sprint_params.sra_avg_time,
codecData->aptx_sprint_params.good_working_level));
KalimbaSendMessage(MESSAGE_SET_APTX_LL_PARAMS1, codecData->aptx_sprint_params.target_codec_level,
codecData->aptx_sprint_params.initial_codec_level,
codecData->aptx_sprint_params.sra_max_rate, /* Third field is scaled by 10000 */
codecData->aptx_sprint_params.sra_avg_time);
KalimbaSendMessage(MESSAGE_SET_APTX_LL_PARAMS2, codecData->aptx_sprint_params.good_working_level,
0, 0, 0);
/* update the current audio state */
SetAudioInUse(TRUE);
}
