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
Plays back a voice prompt once DSP has loaded
*/
static Source csrVoicePresencesGetPrompt(voice_prompt* prompt, koovox_phrase_data_T * pData)
{
const uint8* rx_array;
Source lSource = NULL;
uint16 index;
char file_name[17];
PRINT(("***check_the_source***\n"));
if(!pData)
return NULL;
/* Must be a prompt, work out the index of the prompt */
index = pData->prompt_id;
/* Adjust for language */
index += (pData->language * present_header.no_prompts_per_lang);
PRINT(("***check prompt id***\n"));
/* Sanity checking */
if(index >= present_header.no_prompts || !prompt )
return NULL;
PRINT(("PRESENT: Play prompt %d of %d\n", index+1, present_header.no_prompts));
/* Get the header file name */
sprintf(file_name, "headers/%d.idx", index);
lSource = StreamFileSource(FileFind(FILE_ROOT, file_name, strlen(file_name)));
/* Check source created successfully */
if(SourceSize(lSource) < SIZE_PROMPT_DATA)
{
/* Finished with header source, close it */
SourceClose(lSource);
return NULL;
}
/* Map in header */
rx_array = SourceMap(lSource);
/* Pack data into result */
/* rx_array[0] not used*/
/* rx_array[1] index, not used */
prompt->stereo = rx_array[4];
prompt->size = ((uint32)rx_array[5] << 24) | ((uint32)rx_array[6] << 16) | ((uint16)rx_array[7] << 8) | (rx_array[8]);
prompt->decompression = rx_array[9];
prompt->playback_rate = ((uint16)rx_array[10] << 8) | (rx_array[11]);
/* The size of the prompt must be limited to 16 bits for I2C and SPI as the firmware traps only support a 16 bit size */
if (prompt->size > PRESENT_MAX_SIZE)
{
prompt->size = PRESENT_MAX_SIZE;
PRINT(("Prompt size adjusted to 16 bit maximum\n"));
}
/* Get the prompt file name */
sprintf(file_name, "prompts/%d.prm", index);
PRINT(("File Prompt: %s dec %X rate 0x%x stereo %u size 0x%lx \n", file_name, prompt->decompression,prompt->playback_rate,prompt->stereo,prompt->size));
/* Finished with header source, close it */
if(!SourceClose(lSource))
Panic();
return StreamFileSource(FileFind(FILE_ROOT, file_name, strlen(file_name)));
}
/****************************************************************************
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. */
const char sbc_encoder[] = "sbc_encoder/sbc_encoder.kap";
FILE_INDEX index = FILE_NONE;
typedef struct
{
uint8 content_protection;
uint32 voice_rate;
unsigned bitpool:8;
unsigned format:8;
uint16 packet_size;
uint16 clock_mismatch;
} sbc_codec_data_type;
sbc_codec_data_type *sbc_codecData = (sbc_codec_data_type *) params;
if (!sbc_codecData)
Panic();
index = FileFind(FILE_ROOT, sbc_encoder, sizeof(sbc_encoder)-1);
if (index == FILE_NONE)
Panic();
if (!KalimbaLoad(index))
Panic();
if (!KalimbaSendMessage(KALIMBA_CODEC_TYPE_MESSAGE, 0,0,0,0))
Panic();
CodecSetInputGainNow(codec_task,volume,left_and_right_ch);
SBC = (SBC_t*)PanicUnlessMalloc (sizeof (SBC_t) ) ;
SBC->media_sink[0] = audio_sink ;
SBC->codec_task = codec_task ;
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 */
{
PRINT(("Audio Plugin: SourceAnalog\n"));
/* For analogue input source */
StreamDisconnect(StreamPcmSource(0), StreamPcmSink(0));
StreamDisconnect(StreamPcmSource(1), StreamPcmSink(1));
(void)PcmClearAllRouting();
{
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
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"));
}
