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
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
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
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 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);
}
