| /* $Id$ */ |
| /* |
| * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) |
| * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org> |
| * |
| * This program 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 2 of the License, or |
| * (at your option) any later version. |
| * |
| * This program is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| * GNU General Public License for more details. |
| * |
| * You should have received a copy of the GNU General Public License |
| * along with this program; if not, write to the Free Software |
| * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
| */ |
| #include <pjmedia-audiodev/audiodev_imp.h> |
| #include <pjmedia-audiodev/errno.h> |
| #include <pjmedia/alaw_ulaw.h> |
| #include <pjmedia/resample.h> |
| #include <pjmedia/stereo.h> |
| #include <pj/assert.h> |
| #include <pj/log.h> |
| #include <pj/math.h> |
| #include <pj/os.h> |
| #include <pj/string.h> |
| |
| #if PJMEDIA_AUDIO_DEV_HAS_SYMB_APS |
| |
| #include <e32msgqueue.h> |
| #include <sounddevice.h> |
| #include <APSClientSession.h> |
| #include <pjmedia-codec/amr_helper.h> |
| |
| /* Pack/unpack G.729 frame of S60 DSP codec, taken from: |
| * http://wiki.forum.nokia.com/index.php/TSS000776_-_Payload_conversion_for_G.729_audio_format |
| */ |
| #include "s60_g729_bitstream.h" |
| |
| |
| #define THIS_FILE "symb_aps_dev.c" |
| #define BITS_PER_SAMPLE 16 |
| |
| |
| #if 1 |
| # define TRACE_(st) PJ_LOG(3, st) |
| #else |
| # define TRACE_(st) |
| #endif |
| |
| |
| /* App UID to open global APS queues to communicate with the APS server. */ |
| extern TPtrC APP_UID; |
| |
| /* APS G.711 frame length */ |
| static pj_uint8_t aps_g711_frame_len; |
| |
| |
| /* APS factory */ |
| struct aps_factory |
| { |
| pjmedia_aud_dev_factory base; |
| pj_pool_t *pool; |
| pj_pool_factory *pf; |
| pjmedia_aud_dev_info dev_info; |
| }; |
| |
| |
| /* Forward declaration of CPjAudioEngine */ |
| class CPjAudioEngine; |
| |
| |
| /* APS stream. */ |
| struct aps_stream |
| { |
| // Base |
| pjmedia_aud_stream base; /**< Base class. */ |
| |
| // Pool |
| pj_pool_t *pool; /**< Memory pool. */ |
| |
| // Common settings. |
| pjmedia_aud_param param; /**< Stream param. */ |
| pjmedia_aud_rec_cb rec_cb; /**< Record callback. */ |
| pjmedia_aud_play_cb play_cb; /**< Playback callback. */ |
| void *user_data; /**< Application data. */ |
| |
| // Audio engine |
| CPjAudioEngine *engine; /**< Internal engine. */ |
| |
| pj_timestamp ts_play; /**< Playback timestamp.*/ |
| pj_timestamp ts_rec; /**< Record timestamp. */ |
| |
| pj_int16_t *play_buf; /**< Playback buffer. */ |
| pj_uint16_t play_buf_len; /**< Playback buffer length. */ |
| pj_uint16_t play_buf_start; /**< Playback buffer start index. */ |
| pj_int16_t *rec_buf; /**< Record buffer. */ |
| pj_uint16_t rec_buf_len; /**< Record buffer length. */ |
| void *strm_data; /**< Stream data. */ |
| |
| /* Resampling is needed, in case audio device is opened with clock rate |
| * other than 8kHz (only for PCM format). |
| */ |
| pjmedia_resample *play_resample; /**< Resampler for playback. */ |
| pjmedia_resample *rec_resample; /**< Resampler for recording */ |
| pj_uint16_t resample_factor; /**< Resample factor, requested |
| clock rate / 8000 */ |
| |
| /* When stream is working in PCM format, where the samples may need to be |
| * resampled from/to different clock rate and/or channel count, PCM buffer |
| * is needed to perform such resampling operations. |
| */ |
| pj_int16_t *pcm_buf; /**< PCM buffer. */ |
| }; |
| |
| |
| /* Prototypes */ |
| static pj_status_t factory_init(pjmedia_aud_dev_factory *f); |
| static pj_status_t factory_destroy(pjmedia_aud_dev_factory *f); |
| static pj_status_t factory_refresh(pjmedia_aud_dev_factory *f); |
| static unsigned factory_get_dev_count(pjmedia_aud_dev_factory *f); |
| static pj_status_t factory_get_dev_info(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_dev_info *info); |
| static pj_status_t factory_default_param(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_param *param); |
| static pj_status_t factory_create_stream(pjmedia_aud_dev_factory *f, |
| const pjmedia_aud_param *param, |
| pjmedia_aud_rec_cb rec_cb, |
| pjmedia_aud_play_cb play_cb, |
| void *user_data, |
| pjmedia_aud_stream **p_aud_strm); |
| |
| static pj_status_t stream_get_param(pjmedia_aud_stream *strm, |
| pjmedia_aud_param *param); |
| static pj_status_t stream_get_cap(pjmedia_aud_stream *strm, |
| pjmedia_aud_dev_cap cap, |
| void *value); |
| static pj_status_t stream_set_cap(pjmedia_aud_stream *strm, |
| pjmedia_aud_dev_cap cap, |
| const void *value); |
| static pj_status_t stream_start(pjmedia_aud_stream *strm); |
| static pj_status_t stream_stop(pjmedia_aud_stream *strm); |
| static pj_status_t stream_destroy(pjmedia_aud_stream *strm); |
| |
| |
| /* Operations */ |
| static pjmedia_aud_dev_factory_op factory_op = |
| { |
| &factory_init, |
| &factory_destroy, |
| &factory_get_dev_count, |
| &factory_get_dev_info, |
| &factory_default_param, |
| &factory_create_stream, |
| &factory_refresh |
| }; |
| |
| static pjmedia_aud_stream_op stream_op = |
| { |
| &stream_get_param, |
| &stream_get_cap, |
| &stream_set_cap, |
| &stream_start, |
| &stream_stop, |
| &stream_destroy |
| }; |
| |
| |
| /**************************************************************************** |
| * Internal APS Engine |
| */ |
| |
| /* |
| * Utility: print sound device error |
| */ |
| static void snd_perror(const char *title, TInt rc) |
| { |
| PJ_LOG(1,(THIS_FILE, "%s (error code=%d)", title, rc)); |
| } |
| |
| /* |
| * Utility: wait for specified time. |
| */ |
| static void snd_wait(unsigned ms) |
| { |
| TTime start, now; |
| |
| start.UniversalTime(); |
| do { |
| pj_symbianos_poll(-1, ms); |
| now.UniversalTime(); |
| } while (now.MicroSecondsFrom(start) < ms * 1000); |
| } |
| |
| typedef void(*PjAudioCallback)(TAPSCommBuffer &buf, void *user_data); |
| |
| /** |
| * Abstract class for handler of callbacks from APS client. |
| */ |
| class MQueueHandlerObserver |
| { |
| public: |
| MQueueHandlerObserver(PjAudioCallback RecCb_, PjAudioCallback PlayCb_, |
| void *UserData_) |
| : RecCb(RecCb_), PlayCb(PlayCb_), UserData(UserData_) |
| {} |
| |
| virtual void InputStreamInitialized(const TInt aStatus) = 0; |
| virtual void OutputStreamInitialized(const TInt aStatus) = 0; |
| virtual void NotifyError(const TInt aError) = 0; |
| |
| public: |
| PjAudioCallback RecCb; |
| PjAudioCallback PlayCb; |
| void *UserData; |
| }; |
| |
| /** |
| * Handler for communication and data queue. |
| */ |
| class CQueueHandler : public CActive |
| { |
| public: |
| // Types of queue handler |
| enum TQueueHandlerType { |
| ERecordCommQueue, |
| EPlayCommQueue, |
| ERecordQueue, |
| EPlayQueue |
| }; |
| |
| // The order corresponds to the APS Server state, do not change! |
| enum TState { |
| EAPSPlayerInitialize = 1, |
| EAPSRecorderInitialize = 2, |
| EAPSPlayData = 3, |
| EAPSRecordData = 4, |
| EAPSPlayerInitComplete = 5, |
| EAPSRecorderInitComplete = 6 |
| }; |
| |
| static CQueueHandler* NewL(MQueueHandlerObserver* aObserver, |
| RMsgQueue<TAPSCommBuffer>* aQ, |
| RMsgQueue<TAPSCommBuffer>* aWriteQ, |
| TQueueHandlerType aType) |
| { |
| CQueueHandler* self = new (ELeave) CQueueHandler(aObserver, aQ, aWriteQ, |
| aType); |
| CleanupStack::PushL(self); |
| self->ConstructL(); |
| CleanupStack::Pop(self); |
| return self; |
| } |
| |
| // Destructor |
| ~CQueueHandler() { Cancel(); } |
| |
| // Start listening queue event |
| void Start() { |
| iQ->NotifyDataAvailable(iStatus); |
| SetActive(); |
| } |
| |
| private: |
| // Constructor |
| CQueueHandler(MQueueHandlerObserver* aObserver, |
| RMsgQueue<TAPSCommBuffer>* aQ, |
| RMsgQueue<TAPSCommBuffer>* aWriteQ, |
| TQueueHandlerType aType) |
| : CActive(CActive::EPriorityHigh), |
| iQ(aQ), iWriteQ(aWriteQ), iObserver(aObserver), iType(aType) |
| { |
| CActiveScheduler::Add(this); |
| |
| // use lower priority for comm queues |
| if ((iType == ERecordCommQueue) || (iType == EPlayCommQueue)) |
| SetPriority(CActive::EPriorityStandard); |
| } |
| |
| // Second phase constructor |
| void ConstructL() {} |
| |
| // Inherited from CActive |
| void DoCancel() { iQ->CancelDataAvailable(); } |
| |
| void RunL() { |
| if (iStatus != KErrNone) { |
| iObserver->NotifyError(iStatus.Int()); |
| return; |
| } |
| |
| TAPSCommBuffer buffer; |
| TInt ret = iQ->Receive(buffer); |
| |
| if (ret != KErrNone) { |
| iObserver->NotifyError(ret); |
| return; |
| } |
| |
| switch (iType) { |
| case ERecordQueue: |
| if (buffer.iCommand == EAPSRecordData) { |
| iObserver->RecCb(buffer, iObserver->UserData); |
| } else { |
| iObserver->NotifyError(buffer.iStatus); |
| } |
| break; |
| |
| // Callbacks from the APS main thread |
| case EPlayCommQueue: |
| switch (buffer.iCommand) { |
| case EAPSPlayData: |
| if (buffer.iStatus == KErrUnderflow) { |
| iObserver->PlayCb(buffer, iObserver->UserData); |
| iWriteQ->Send(buffer); |
| } |
| break; |
| case EAPSPlayerInitialize: |
| iObserver->NotifyError(buffer.iStatus); |
| break; |
| case EAPSPlayerInitComplete: |
| iObserver->OutputStreamInitialized(buffer.iStatus); |
| break; |
| case EAPSRecorderInitComplete: |
| iObserver->InputStreamInitialized(buffer.iStatus); |
| break; |
| default: |
| iObserver->NotifyError(buffer.iStatus); |
| break; |
| } |
| break; |
| |
| // Callbacks from the APS recorder thread |
| case ERecordCommQueue: |
| switch (buffer.iCommand) { |
| // The APS recorder thread will only report errors |
| // through this handler. All other callbacks will be |
| // sent from the APS main thread through EPlayCommQueue |
| case EAPSRecorderInitialize: |
| case EAPSRecordData: |
| default: |
| iObserver->NotifyError(buffer.iStatus); |
| break; |
| } |
| break; |
| |
| default: |
| break; |
| } |
| |
| // issue next request |
| iQ->NotifyDataAvailable(iStatus); |
| SetActive(); |
| } |
| |
| TInt RunError(TInt) { |
| return 0; |
| } |
| |
| // Data |
| RMsgQueue<TAPSCommBuffer> *iQ; // (not owned) |
| RMsgQueue<TAPSCommBuffer> *iWriteQ; // (not owned) |
| MQueueHandlerObserver *iObserver; // (not owned) |
| TQueueHandlerType iType; |
| }; |
| |
| /* |
| * Audio setting for CPjAudioEngine. |
| */ |
| class CPjAudioSetting |
| { |
| public: |
| TFourCC fourcc; |
| TAPSCodecMode mode; |
| TBool plc; |
| TBool vad; |
| TBool cng; |
| TBool loudspk; |
| }; |
| |
| /* |
| * Implementation: Symbian Input & Output Stream. |
| */ |
| class CPjAudioEngine : public CBase, MQueueHandlerObserver |
| { |
| public: |
| enum State |
| { |
| STATE_NULL, |
| STATE_INITIALIZING, |
| STATE_READY, |
| STATE_STREAMING, |
| STATE_PENDING_STOP |
| }; |
| |
| ~CPjAudioEngine(); |
| |
| static CPjAudioEngine *NewL(struct aps_stream *parent_strm, |
| PjAudioCallback rec_cb, |
| PjAudioCallback play_cb, |
| void *user_data, |
| const CPjAudioSetting &setting); |
| |
| TInt StartL(); |
| void Stop(); |
| |
| TInt ActivateSpeaker(TBool active); |
| |
| TInt SetVolume(TInt vol) { return iSession.SetVolume(vol); } |
| TInt GetVolume() { return iSession.Volume(); } |
| TInt GetMaxVolume() { return iSession.MaxVolume(); } |
| |
| TInt SetGain(TInt gain) { return iSession.SetGain(gain); } |
| TInt GetGain() { return iSession.Gain(); } |
| TInt GetMaxGain() { return iSession.MaxGain(); } |
| |
| private: |
| CPjAudioEngine(struct aps_stream *parent_strm, |
| PjAudioCallback rec_cb, |
| PjAudioCallback play_cb, |
| void *user_data, |
| const CPjAudioSetting &setting); |
| void ConstructL(); |
| |
| TInt InitPlayL(); |
| TInt InitRecL(); |
| TInt StartStreamL(); |
| void Deinit(); |
| |
| // Inherited from MQueueHandlerObserver |
| virtual void InputStreamInitialized(const TInt aStatus); |
| virtual void OutputStreamInitialized(const TInt aStatus); |
| virtual void NotifyError(const TInt aError); |
| |
| TBool session_opened; |
| State state_; |
| struct aps_stream *parentStrm_; |
| CPjAudioSetting setting_; |
| |
| RAPSSession iSession; |
| TAPSInitSettings iPlaySettings; |
| TAPSInitSettings iRecSettings; |
| |
| RMsgQueue<TAPSCommBuffer> iReadQ; |
| RMsgQueue<TAPSCommBuffer> iReadCommQ; |
| TBool readq_opened; |
| RMsgQueue<TAPSCommBuffer> iWriteQ; |
| RMsgQueue<TAPSCommBuffer> iWriteCommQ; |
| TBool writeq_opened; |
| |
| CQueueHandler *iPlayCommHandler; |
| CQueueHandler *iRecCommHandler; |
| CQueueHandler *iRecHandler; |
| }; |
| |
| |
| CPjAudioEngine* CPjAudioEngine::NewL(struct aps_stream *parent_strm, |
| PjAudioCallback rec_cb, |
| PjAudioCallback play_cb, |
| void *user_data, |
| const CPjAudioSetting &setting) |
| { |
| CPjAudioEngine* self = new (ELeave) CPjAudioEngine(parent_strm, |
| rec_cb, play_cb, |
| user_data, |
| setting); |
| CleanupStack::PushL(self); |
| self->ConstructL(); |
| CleanupStack::Pop(self); |
| return self; |
| } |
| |
| CPjAudioEngine::CPjAudioEngine(struct aps_stream *parent_strm, |
| PjAudioCallback rec_cb, |
| PjAudioCallback play_cb, |
| void *user_data, |
| const CPjAudioSetting &setting) |
| : MQueueHandlerObserver(rec_cb, play_cb, user_data), |
| session_opened(EFalse), |
| state_(STATE_NULL), |
| parentStrm_(parent_strm), |
| setting_(setting), |
| readq_opened(EFalse), |
| writeq_opened(EFalse), |
| iPlayCommHandler(0), |
| iRecCommHandler(0), |
| iRecHandler(0) |
| { |
| } |
| |
| CPjAudioEngine::~CPjAudioEngine() |
| { |
| Deinit(); |
| |
| TRACE_((THIS_FILE, "Sound device destroyed")); |
| } |
| |
| TInt CPjAudioEngine::InitPlayL() |
| { |
| TInt err = iSession.InitializePlayer(iPlaySettings); |
| if (err != KErrNone) { |
| Deinit(); |
| snd_perror("Failed to initialize player", err); |
| return err; |
| } |
| |
| // Open message queues for the output stream |
| TBuf<128> buf2 = iPlaySettings.iGlobal; |
| buf2.Append(_L("PlayQueue")); |
| TBuf<128> buf3 = iPlaySettings.iGlobal; |
| buf3.Append(_L("PlayCommQueue")); |
| |
| while (iWriteQ.OpenGlobal(buf2)) |
| User::After(10); |
| while (iWriteCommQ.OpenGlobal(buf3)) |
| User::After(10); |
| |
| writeq_opened = ETrue; |
| |
| // Construct message queue handler |
| iPlayCommHandler = CQueueHandler::NewL(this, &iWriteCommQ, &iWriteQ, |
| CQueueHandler::EPlayCommQueue); |
| |
| // Start observing APS callbacks on output stream message queue |
| iPlayCommHandler->Start(); |
| |
| return 0; |
| } |
| |
| TInt CPjAudioEngine::InitRecL() |
| { |
| // Initialize input stream device |
| TInt err = iSession.InitializeRecorder(iRecSettings); |
| if (err != KErrNone && err != KErrAlreadyExists) { |
| Deinit(); |
| snd_perror("Failed to initialize recorder", err); |
| return err; |
| } |
| |
| TBuf<128> buf1 = iRecSettings.iGlobal; |
| buf1.Append(_L("RecordQueue")); |
| TBuf<128> buf4 = iRecSettings.iGlobal; |
| buf4.Append(_L("RecordCommQueue")); |
| |
| // Must wait for APS thread to finish creating message queues |
| // before we can open and use them. |
| while (iReadQ.OpenGlobal(buf1)) |
| User::After(10); |
| while (iReadCommQ.OpenGlobal(buf4)) |
| User::After(10); |
| |
| readq_opened = ETrue; |
| |
| // Construct message queue handlers |
| iRecHandler = CQueueHandler::NewL(this, &iReadQ, NULL, |
| CQueueHandler::ERecordQueue); |
| iRecCommHandler = CQueueHandler::NewL(this, &iReadCommQ, NULL, |
| CQueueHandler::ERecordCommQueue); |
| |
| // Start observing APS callbacks from on input stream message queue |
| iRecHandler->Start(); |
| iRecCommHandler->Start(); |
| |
| return 0; |
| } |
| |
| TInt CPjAudioEngine::StartL() |
| { |
| if (state_ == STATE_READY) |
| return StartStreamL(); |
| |
| PJ_ASSERT_RETURN(state_ == STATE_NULL, PJMEDIA_EAUD_INVOP); |
| |
| if (!session_opened) { |
| TInt err = iSession.Connect(); |
| if (err != KErrNone) |
| return err; |
| session_opened = ETrue; |
| } |
| |
| // Even if only capturer are opened, playback thread of APS Server need |
| // to be run(?). Since some messages will be delivered via play comm queue. |
| state_ = STATE_INITIALIZING; |
| |
| return InitPlayL(); |
| } |
| |
| void CPjAudioEngine::Stop() |
| { |
| if (state_ == STATE_STREAMING) { |
| iSession.Stop(); |
| state_ = STATE_READY; |
| TRACE_((THIS_FILE, "Sound device stopped")); |
| } else if (state_ == STATE_INITIALIZING) { |
| // Initialization is on progress, so let's set the state to |
| // STATE_PENDING_STOP to prevent it starting the stream. |
| state_ = STATE_PENDING_STOP; |
| |
| // Then wait until initialization done. |
| while (state_ != STATE_READY && state_ != STATE_NULL) |
| pj_symbianos_poll(-1, 100); |
| } |
| } |
| |
| void CPjAudioEngine::ConstructL() |
| { |
| // Recorder settings |
| iRecSettings.iFourCC = setting_.fourcc; |
| iRecSettings.iGlobal = APP_UID; |
| iRecSettings.iPriority = TMdaPriority(100); |
| iRecSettings.iPreference = TMdaPriorityPreference(0x05210001); |
| iRecSettings.iSettings.iChannels = EMMFMono; |
| iRecSettings.iSettings.iSampleRate = EMMFSampleRate8000Hz; |
| |
| // Player settings |
| iPlaySettings.iFourCC = setting_.fourcc; |
| iPlaySettings.iGlobal = APP_UID; |
| iPlaySettings.iPriority = TMdaPriority(100); |
| iPlaySettings.iPreference = TMdaPriorityPreference(0x05220001); |
| iPlaySettings.iSettings.iChannels = EMMFMono; |
| iPlaySettings.iSettings.iSampleRate = EMMFSampleRate8000Hz; |
| iPlaySettings.iSettings.iVolume = 0; |
| |
| User::LeaveIfError(iSession.Connect()); |
| session_opened = ETrue; |
| } |
| |
| TInt CPjAudioEngine::StartStreamL() |
| { |
| pj_assert(state_==STATE_READY || state_==STATE_INITIALIZING); |
| |
| iSession.SetCng(setting_.cng); |
| iSession.SetVadMode(setting_.vad); |
| iSession.SetPlc(setting_.plc); |
| iSession.SetEncoderMode(setting_.mode); |
| iSession.SetDecoderMode(setting_.mode); |
| iSession.ActivateLoudspeaker(setting_.loudspk); |
| |
| // Not only capture |
| if (parentStrm_->param.dir != PJMEDIA_DIR_CAPTURE) { |
| iSession.Write(); |
| TRACE_((THIS_FILE, "Player started")); |
| } |
| |
| // Not only playback |
| if (parentStrm_->param.dir != PJMEDIA_DIR_PLAYBACK) { |
| iSession.Read(); |
| TRACE_((THIS_FILE, "Recorder started")); |
| } |
| |
| state_ = STATE_STREAMING; |
| |
| return 0; |
| } |
| |
| void CPjAudioEngine::Deinit() |
| { |
| Stop(); |
| |
| delete iRecHandler; |
| delete iPlayCommHandler; |
| delete iRecCommHandler; |
| |
| if (session_opened) { |
| enum { APS_CLOSE_WAIT_TIME = 200 }; /* in msecs */ |
| |
| // On some devices, immediate closing after stopping may cause |
| // APS server panic KERN-EXEC 0, so let's wait for sometime before |
| // closing the client session. |
| snd_wait(APS_CLOSE_WAIT_TIME); |
| |
| iSession.Close(); |
| session_opened = EFalse; |
| } |
| |
| if (readq_opened) { |
| iReadQ.Close(); |
| iReadCommQ.Close(); |
| readq_opened = EFalse; |
| } |
| |
| if (writeq_opened) { |
| iWriteQ.Close(); |
| iWriteCommQ.Close(); |
| writeq_opened = EFalse; |
| } |
| |
| state_ = STATE_NULL; |
| } |
| |
| void CPjAudioEngine::InputStreamInitialized(const TInt aStatus) |
| { |
| TRACE_((THIS_FILE, "Recorder initialized, err=%d", aStatus)); |
| |
| if (aStatus == KErrNone) { |
| // Don't start the stream since Stop() has been requested. |
| if (state_ != STATE_PENDING_STOP) { |
| StartStreamL(); |
| } else { |
| state_ = STATE_READY; |
| } |
| } else { |
| Deinit(); |
| } |
| } |
| |
| void CPjAudioEngine::OutputStreamInitialized(const TInt aStatus) |
| { |
| TRACE_((THIS_FILE, "Player initialized, err=%d", aStatus)); |
| |
| if (aStatus == KErrNone) { |
| if (parentStrm_->param.dir == PJMEDIA_DIR_PLAYBACK) { |
| // Don't start the stream since Stop() has been requested. |
| if (state_ != STATE_PENDING_STOP) { |
| StartStreamL(); |
| } else { |
| state_ = STATE_READY; |
| } |
| } else |
| InitRecL(); |
| } else { |
| Deinit(); |
| } |
| } |
| |
| void CPjAudioEngine::NotifyError(const TInt aError) |
| { |
| Deinit(); |
| snd_perror("Error from CQueueHandler", aError); |
| } |
| |
| TInt CPjAudioEngine::ActivateSpeaker(TBool active) |
| { |
| if (state_ == STATE_READY || state_ == STATE_STREAMING) { |
| iSession.ActivateLoudspeaker(active); |
| TRACE_((THIS_FILE, "Loudspeaker turned %s", (active? "on":"off"))); |
| return KErrNone; |
| } |
| return KErrNotReady; |
| } |
| |
| /**************************************************************************** |
| * Internal APS callbacks for PCM format |
| */ |
| |
| static void RecCbPcm(TAPSCommBuffer &buf, void *user_data) |
| { |
| struct aps_stream *strm = (struct aps_stream*) user_data; |
| |
| /* Buffer has to contain normal speech. */ |
| pj_assert(buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0); |
| |
| /* Detect the recorder G.711 frame size, player frame size will follow |
| * this recorder frame size. |
| */ |
| if (aps_g711_frame_len == 0) { |
| aps_g711_frame_len = buf.iBuffer.Length() < 160? 80 : 160; |
| TRACE_((THIS_FILE, "Detected APS G.711 frame size = %u samples", |
| aps_g711_frame_len)); |
| } |
| |
| /* Decode APS buffer (coded in G.711) and put the PCM result into rec_buf. |
| * Whenever rec_buf is full, call parent stream callback. |
| */ |
| unsigned samples_processed = 0; |
| |
| while (samples_processed < aps_g711_frame_len) { |
| unsigned samples_to_process; |
| unsigned samples_req; |
| |
| samples_to_process = aps_g711_frame_len - samples_processed; |
| samples_req = (strm->param.samples_per_frame / |
| strm->param.channel_count / |
| strm->resample_factor) - |
| strm->rec_buf_len; |
| if (samples_to_process > samples_req) |
| samples_to_process = samples_req; |
| |
| pjmedia_ulaw_decode(&strm->rec_buf[strm->rec_buf_len], |
| buf.iBuffer.Ptr() + 2 + samples_processed, |
| samples_to_process); |
| |
| strm->rec_buf_len += samples_to_process; |
| samples_processed += samples_to_process; |
| |
| /* Buffer is full, time to call parent callback */ |
| if (strm->rec_buf_len == strm->param.samples_per_frame / |
| strm->param.channel_count / |
| strm->resample_factor) |
| { |
| pjmedia_frame f; |
| |
| /* Need to resample clock rate? */ |
| if (strm->rec_resample) { |
| unsigned resampled = 0; |
| |
| while (resampled < strm->rec_buf_len) { |
| pjmedia_resample_run(strm->rec_resample, |
| &strm->rec_buf[resampled], |
| strm->pcm_buf + |
| resampled * strm->resample_factor); |
| resampled += 80; |
| } |
| f.buf = strm->pcm_buf; |
| } else { |
| f.buf = strm->rec_buf; |
| } |
| |
| /* Need to convert channel count? */ |
| if (strm->param.channel_count != 1) { |
| pjmedia_convert_channel_1ton((pj_int16_t*)f.buf, |
| (pj_int16_t*)f.buf, |
| strm->param.channel_count, |
| strm->param.samples_per_frame / |
| strm->param.channel_count, |
| 0); |
| } |
| |
| /* Call parent callback */ |
| f.type = PJMEDIA_FRAME_TYPE_AUDIO; |
| f.size = strm->param.samples_per_frame << 1; |
| strm->rec_cb(strm->user_data, &f); |
| strm->rec_buf_len = 0; |
| } |
| } |
| } |
| |
| static void PlayCbPcm(TAPSCommBuffer &buf, void *user_data) |
| { |
| struct aps_stream *strm = (struct aps_stream*) user_data; |
| unsigned g711_frame_len = aps_g711_frame_len; |
| |
| /* Init buffer attributes and header. */ |
| buf.iCommand = CQueueHandler::EAPSPlayData; |
| buf.iStatus = 0; |
| buf.iBuffer.Zero(); |
| buf.iBuffer.Append(1); |
| buf.iBuffer.Append(0); |
| |
| /* Assume frame size is 10ms if frame size hasn't been known. */ |
| if (g711_frame_len == 0) |
| g711_frame_len = 80; |
| |
| /* Call parent stream callback to get PCM samples to play, |
| * encode the PCM samples into G.711 and put it into APS buffer. |
| */ |
| unsigned samples_processed = 0; |
| |
| while (samples_processed < g711_frame_len) { |
| /* Need more samples to play, time to call parent callback */ |
| if (strm->play_buf_len == 0) { |
| pjmedia_frame f; |
| unsigned samples_got; |
| |
| f.size = strm->param.samples_per_frame << 1; |
| if (strm->play_resample || strm->param.channel_count != 1) |
| f.buf = strm->pcm_buf; |
| else |
| f.buf = strm->play_buf; |
| |
| /* Call parent callback */ |
| strm->play_cb(strm->user_data, &f); |
| if (f.type != PJMEDIA_FRAME_TYPE_AUDIO) { |
| pjmedia_zero_samples((pj_int16_t*)f.buf, |
| strm->param.samples_per_frame); |
| } |
| |
| samples_got = strm->param.samples_per_frame / |
| strm->param.channel_count / |
| strm->resample_factor; |
| |
| /* Need to convert channel count? */ |
| if (strm->param.channel_count != 1) { |
| pjmedia_convert_channel_nto1((pj_int16_t*)f.buf, |
| (pj_int16_t*)f.buf, |
| strm->param.channel_count, |
| strm->param.samples_per_frame, |
| PJ_FALSE, |
| 0); |
| } |
| |
| /* Need to resample clock rate? */ |
| if (strm->play_resample) { |
| unsigned resampled = 0; |
| |
| while (resampled < samples_got) |
| { |
| pjmedia_resample_run(strm->play_resample, |
| strm->pcm_buf + |
| resampled * strm->resample_factor, |
| &strm->play_buf[resampled]); |
| resampled += 80; |
| } |
| } |
| |
| strm->play_buf_len = samples_got; |
| strm->play_buf_start = 0; |
| } |
| |
| unsigned tmp; |
| |
| tmp = PJ_MIN(strm->play_buf_len, g711_frame_len - samples_processed); |
| pjmedia_ulaw_encode((pj_uint8_t*)&strm->play_buf[strm->play_buf_start], |
| &strm->play_buf[strm->play_buf_start], |
| tmp); |
| buf.iBuffer.Append((TUint8*)&strm->play_buf[strm->play_buf_start], tmp); |
| samples_processed += tmp; |
| strm->play_buf_len -= tmp; |
| strm->play_buf_start += tmp; |
| } |
| } |
| |
| /**************************************************************************** |
| * Internal APS callbacks for non-PCM format |
| */ |
| |
| static void RecCb(TAPSCommBuffer &buf, void *user_data) |
| { |
| struct aps_stream *strm = (struct aps_stream*) user_data; |
| pjmedia_frame_ext *frame = (pjmedia_frame_ext*) strm->rec_buf; |
| |
| switch(strm->param.ext_fmt.id) { |
| case PJMEDIA_FORMAT_AMR: |
| { |
| const pj_uint8_t *p = (const pj_uint8_t*)buf.iBuffer.Ptr() + 1; |
| unsigned len = buf.iBuffer.Length() - 1; |
| |
| pjmedia_frame_ext_append_subframe(frame, p, len << 3, 160); |
| if (frame->samples_cnt == strm->param.samples_per_frame) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->rec_cb(strm->user_data, (pjmedia_frame*)frame); |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_G729: |
| { |
| /* Check if we got a normal or SID frame. */ |
| if (buf.iBuffer[0] != 0 || buf.iBuffer[1] != 0) { |
| enum { NORMAL_LEN = 22, SID_LEN = 8 }; |
| TBitStream *bitstream = (TBitStream*)strm->strm_data; |
| unsigned src_len = buf.iBuffer.Length()- 2; |
| |
| pj_assert(src_len == NORMAL_LEN || src_len == SID_LEN); |
| |
| const TDesC8& p = bitstream->CompressG729Frame( |
| buf.iBuffer.Right(src_len), |
| src_len == SID_LEN); |
| |
| pjmedia_frame_ext_append_subframe(frame, p.Ptr(), |
| p.Length() << 3, 80); |
| } else { /* We got null frame. */ |
| pjmedia_frame_ext_append_subframe(frame, NULL, 0, 80); |
| } |
| |
| if (frame->samples_cnt == strm->param.samples_per_frame) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->rec_cb(strm->user_data, (pjmedia_frame*)frame); |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_ILBC: |
| { |
| unsigned samples_got; |
| |
| samples_got = |
| strm->param.ext_fmt.det.aud.avg_bps == 15200? 160 : 240; |
| |
| /* Check if we got a normal frame. */ |
| if (buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0) { |
| const pj_uint8_t *p = (const pj_uint8_t*)buf.iBuffer.Ptr() + 2; |
| unsigned len = buf.iBuffer.Length() - 2; |
| |
| pjmedia_frame_ext_append_subframe(frame, p, len << 3, |
| samples_got); |
| } else { /* We got null frame. */ |
| pjmedia_frame_ext_append_subframe(frame, NULL, 0, samples_got); |
| } |
| |
| if (frame->samples_cnt == strm->param.samples_per_frame) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->rec_cb(strm->user_data, (pjmedia_frame*)frame); |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_PCMU: |
| case PJMEDIA_FORMAT_PCMA: |
| { |
| unsigned samples_processed = 0; |
| |
| /* Make sure it is normal frame. */ |
| pj_assert(buf.iBuffer[0] == 1 && buf.iBuffer[1] == 0); |
| |
| /* Detect the recorder G.711 frame size, player frame size will |
| * follow this recorder frame size. |
| */ |
| if (aps_g711_frame_len == 0) { |
| aps_g711_frame_len = buf.iBuffer.Length() < 160? 80 : 160; |
| TRACE_((THIS_FILE, "Detected APS G.711 frame size = %u samples", |
| aps_g711_frame_len)); |
| } |
| |
| /* Convert APS buffer format into pjmedia_frame_ext. Whenever |
| * samples count in the frame is equal to stream's samples per |
| * frame, call parent stream callback. |
| */ |
| while (samples_processed < aps_g711_frame_len) { |
| unsigned tmp; |
| const pj_uint8_t *pb = (const pj_uint8_t*)buf.iBuffer.Ptr() + |
| 2 + samples_processed; |
| |
| tmp = PJ_MIN(strm->param.samples_per_frame - frame->samples_cnt, |
| aps_g711_frame_len - samples_processed); |
| |
| pjmedia_frame_ext_append_subframe(frame, pb, tmp << 3, tmp); |
| samples_processed += tmp; |
| |
| if (frame->samples_cnt == strm->param.samples_per_frame) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->rec_cb(strm->user_data, (pjmedia_frame*)frame); |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| } |
| break; |
| |
| default: |
| break; |
| } |
| } |
| |
| static void PlayCb(TAPSCommBuffer &buf, void *user_data) |
| { |
| struct aps_stream *strm = (struct aps_stream*) user_data; |
| pjmedia_frame_ext *frame = (pjmedia_frame_ext*) strm->play_buf; |
| |
| /* Init buffer attributes and header. */ |
| buf.iCommand = CQueueHandler::EAPSPlayData; |
| buf.iStatus = 0; |
| buf.iBuffer.Zero(); |
| |
| switch(strm->param.ext_fmt.id) { |
| case PJMEDIA_FORMAT_AMR: |
| { |
| if (frame->samples_cnt == 0) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->play_cb(strm->user_data, (pjmedia_frame*)frame); |
| pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED || |
| frame->base.type==PJMEDIA_FRAME_TYPE_NONE); |
| } |
| |
| if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { |
| pjmedia_frame_ext_subframe *sf; |
| unsigned samples_cnt; |
| |
| sf = pjmedia_frame_ext_get_subframe(frame, 0); |
| samples_cnt = frame->samples_cnt / frame->subframe_cnt; |
| |
| if (sf->data && sf->bitlen) { |
| /* AMR header for APS is one byte, the format (may be!): |
| * 0xxxxy00, where xxxx:frame type, y:not sure. |
| */ |
| unsigned len = (sf->bitlen+7)>>3; |
| enum {SID_FT = 8 }; |
| pj_uint8_t amr_header = 4, ft = SID_FT; |
| |
| if (len >= pjmedia_codec_amrnb_framelen[0]) |
| ft = pjmedia_codec_amr_get_mode2(PJ_TRUE, len); |
| |
| amr_header |= ft << 3; |
| buf.iBuffer.Append(amr_header); |
| |
| buf.iBuffer.Append((TUint8*)sf->data, len); |
| } else { |
| enum {NO_DATA_FT = 15 }; |
| pj_uint8_t amr_header = 4 | (NO_DATA_FT << 3); |
| |
| buf.iBuffer.Append(amr_header); |
| } |
| |
| pjmedia_frame_ext_pop_subframes(frame, 1); |
| |
| } else { /* PJMEDIA_FRAME_TYPE_NONE */ |
| enum {NO_DATA_FT = 15 }; |
| pj_uint8_t amr_header = 4 | (NO_DATA_FT << 3); |
| |
| buf.iBuffer.Append(amr_header); |
| |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_G729: |
| { |
| if (frame->samples_cnt == 0) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->play_cb(strm->user_data, (pjmedia_frame*)frame); |
| pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED || |
| frame->base.type==PJMEDIA_FRAME_TYPE_NONE); |
| } |
| |
| if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { |
| pjmedia_frame_ext_subframe *sf; |
| unsigned samples_cnt; |
| |
| sf = pjmedia_frame_ext_get_subframe(frame, 0); |
| samples_cnt = frame->samples_cnt / frame->subframe_cnt; |
| |
| if (sf->data && sf->bitlen) { |
| enum { NORMAL_LEN = 10, SID_LEN = 2 }; |
| pj_bool_t sid_frame = ((sf->bitlen >> 3) == SID_LEN); |
| TBitStream *bitstream = (TBitStream*)strm->strm_data; |
| const TPtrC8 src(sf->data, sf->bitlen>>3); |
| const TDesC8 &dst = bitstream->ExpandG729Frame(src, |
| sid_frame); |
| if (sid_frame) { |
| buf.iBuffer.Append(2); |
| buf.iBuffer.Append(0); |
| } else { |
| buf.iBuffer.Append(1); |
| buf.iBuffer.Append(0); |
| } |
| buf.iBuffer.Append(dst); |
| } else { |
| buf.iBuffer.Append(2); |
| buf.iBuffer.Append(0); |
| buf.iBuffer.AppendFill(0, 22); |
| } |
| |
| pjmedia_frame_ext_pop_subframes(frame, 1); |
| |
| } else { /* PJMEDIA_FRAME_TYPE_NONE */ |
| buf.iBuffer.Append(2); |
| buf.iBuffer.Append(0); |
| buf.iBuffer.AppendFill(0, 22); |
| |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_ILBC: |
| { |
| if (frame->samples_cnt == 0) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->play_cb(strm->user_data, (pjmedia_frame*)frame); |
| pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED || |
| frame->base.type==PJMEDIA_FRAME_TYPE_NONE); |
| } |
| |
| if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { |
| pjmedia_frame_ext_subframe *sf; |
| unsigned samples_cnt; |
| |
| sf = pjmedia_frame_ext_get_subframe(frame, 0); |
| samples_cnt = frame->samples_cnt / frame->subframe_cnt; |
| |
| pj_assert((strm->param.ext_fmt.det.aud.avg_bps == 15200 && |
| samples_cnt == 160) || |
| (strm->param.ext_fmt.det.aud.avg_bps != 15200 && |
| samples_cnt == 240)); |
| |
| if (sf->data && sf->bitlen) { |
| buf.iBuffer.Append(1); |
| buf.iBuffer.Append(0); |
| buf.iBuffer.Append((TUint8*)sf->data, sf->bitlen>>3); |
| } else { |
| buf.iBuffer.Append(0); |
| buf.iBuffer.Append(0); |
| } |
| |
| pjmedia_frame_ext_pop_subframes(frame, 1); |
| |
| } else { /* PJMEDIA_FRAME_TYPE_NONE */ |
| buf.iBuffer.Append(0); |
| buf.iBuffer.Append(0); |
| |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| break; |
| |
| case PJMEDIA_FORMAT_PCMU: |
| case PJMEDIA_FORMAT_PCMA: |
| { |
| unsigned samples_ready = 0; |
| unsigned samples_req = aps_g711_frame_len; |
| |
| /* Assume frame size is 10ms if frame size hasn't been known. */ |
| if (samples_req == 0) |
| samples_req = 80; |
| |
| buf.iBuffer.Append(1); |
| buf.iBuffer.Append(0); |
| |
| /* Call parent stream callback to get samples to play. */ |
| while (samples_ready < samples_req) { |
| if (frame->samples_cnt == 0) { |
| frame->base.type = PJMEDIA_FRAME_TYPE_EXTENDED; |
| strm->play_cb(strm->user_data, (pjmedia_frame*)frame); |
| pj_assert(frame->base.type==PJMEDIA_FRAME_TYPE_EXTENDED || |
| frame->base.type==PJMEDIA_FRAME_TYPE_NONE); |
| } |
| |
| if (frame->base.type == PJMEDIA_FRAME_TYPE_EXTENDED) { |
| pjmedia_frame_ext_subframe *sf; |
| unsigned samples_cnt; |
| |
| sf = pjmedia_frame_ext_get_subframe(frame, 0); |
| samples_cnt = frame->samples_cnt / frame->subframe_cnt; |
| if (sf->data && sf->bitlen) { |
| buf.iBuffer.Append((TUint8*)sf->data, sf->bitlen>>3); |
| } else { |
| pj_uint8_t silc; |
| silc = (strm->param.ext_fmt.id==PJMEDIA_FORMAT_PCMU)? |
| pjmedia_linear2ulaw(0) : pjmedia_linear2alaw(0); |
| buf.iBuffer.AppendFill(silc, samples_cnt); |
| } |
| samples_ready += samples_cnt; |
| |
| pjmedia_frame_ext_pop_subframes(frame, 1); |
| |
| } else { /* PJMEDIA_FRAME_TYPE_NONE */ |
| pj_uint8_t silc; |
| |
| silc = (strm->param.ext_fmt.id==PJMEDIA_FORMAT_PCMU)? |
| pjmedia_linear2ulaw(0) : pjmedia_linear2alaw(0); |
| buf.iBuffer.AppendFill(silc, samples_req - samples_ready); |
| |
| samples_ready = samples_req; |
| frame->samples_cnt = 0; |
| frame->subframe_cnt = 0; |
| } |
| } |
| } |
| break; |
| |
| default: |
| break; |
| } |
| } |
| |
| |
| /**************************************************************************** |
| * Factory operations |
| */ |
| |
| /* |
| * C compatible declaration of APS factory. |
| */ |
| PJ_BEGIN_DECL |
| PJ_DECL(pjmedia_aud_dev_factory*) pjmedia_aps_factory(pj_pool_factory *pf); |
| PJ_END_DECL |
| |
| /* |
| * Init APS audio driver. |
| */ |
| PJ_DEF(pjmedia_aud_dev_factory*) pjmedia_aps_factory(pj_pool_factory *pf) |
| { |
| struct aps_factory *f; |
| pj_pool_t *pool; |
| |
| pool = pj_pool_create(pf, "APS", 1000, 1000, NULL); |
| f = PJ_POOL_ZALLOC_T(pool, struct aps_factory); |
| f->pf = pf; |
| f->pool = pool; |
| f->base.op = &factory_op; |
| |
| return &f->base; |
| } |
| |
| /* API: init factory */ |
| static pj_status_t factory_init(pjmedia_aud_dev_factory *f) |
| { |
| struct aps_factory *af = (struct aps_factory*)f; |
| |
| pj_ansi_strcpy(af->dev_info.name, "S60 APS"); |
| af->dev_info.default_samples_per_sec = 8000; |
| af->dev_info.caps = PJMEDIA_AUD_DEV_CAP_EXT_FORMAT | |
| //PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING | |
| PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING | |
| PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE | |
| PJMEDIA_AUD_DEV_CAP_VAD | |
| PJMEDIA_AUD_DEV_CAP_CNG; |
| af->dev_info.routes = PJMEDIA_AUD_DEV_ROUTE_EARPIECE | |
| PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER; |
| af->dev_info.input_count = 1; |
| af->dev_info.output_count = 1; |
| |
| /* Enumerate codecs by trying to initialize each codec and examining |
| * the error code. Consider the following: |
| * - not possible to reinitialize the same APS session with |
| * different settings, |
| * - closing APS session and trying to immediately reconnect may fail, |
| * clients should wait ~5s before attempting to reconnect. |
| */ |
| |
| unsigned i, fmt_cnt = 0; |
| pj_bool_t g711_supported = PJ_FALSE; |
| |
| /* Do not change the order! */ |
| TFourCC fourcc[] = { |
| TFourCC(KMCPFourCCIdAMRNB), |
| TFourCC(KMCPFourCCIdG711), |
| TFourCC(KMCPFourCCIdG729), |
| TFourCC(KMCPFourCCIdILBC) |
| }; |
| |
| for (i = 0; i < PJ_ARRAY_SIZE(fourcc); ++i) { |
| pj_bool_t supported = PJ_FALSE; |
| unsigned retry_cnt = 0; |
| enum { MAX_RETRY = 3 }; |
| |
| #if (PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC == 0) |
| /* Codec detection is disabled */ |
| supported = PJ_TRUE; |
| #elif (PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC == 1) |
| /* Minimal codec detection, AMR-NB and G.711 only */ |
| if (i > 1) { |
| /* If G.711 has been checked, skip G.729 and iLBC checks */ |
| retry_cnt = MAX_RETRY; |
| supported = g711_supported; |
| } |
| #endif |
| |
| while (!supported && ++retry_cnt <= MAX_RETRY) { |
| RAPSSession iSession; |
| TAPSInitSettings iPlaySettings; |
| TAPSInitSettings iRecSettings; |
| TInt err; |
| |
| // Recorder settings |
| iRecSettings.iGlobal = APP_UID; |
| iRecSettings.iPriority = TMdaPriority(100); |
| iRecSettings.iPreference = TMdaPriorityPreference(0x05210001); |
| iRecSettings.iSettings.iChannels = EMMFMono; |
| iRecSettings.iSettings.iSampleRate = EMMFSampleRate8000Hz; |
| |
| // Player settings |
| iPlaySettings.iGlobal = APP_UID; |
| iPlaySettings.iPriority = TMdaPriority(100); |
| iPlaySettings.iPreference = TMdaPriorityPreference(0x05220001); |
| iPlaySettings.iSettings.iChannels = EMMFMono; |
| iPlaySettings.iSettings.iSampleRate = EMMFSampleRate8000Hz; |
| |
| iRecSettings.iFourCC = iPlaySettings.iFourCC = fourcc[i]; |
| |
| err = iSession.Connect(); |
| if (err == KErrNone) |
| err = iSession.InitializePlayer(iPlaySettings); |
| if (err == KErrNone) |
| err = iSession.InitializeRecorder(iRecSettings); |
| |
| // On some devices, immediate closing causes APS Server panic, |
| // e.g: N95, so let's just wait for some time before closing. |
| enum { APS_CLOSE_WAIT_TIME = 200 }; /* in msecs */ |
| snd_wait(APS_CLOSE_WAIT_TIME); |
| |
| iSession.Close(); |
| |
| if (err == KErrNone) { |
| /* All fine, stop retyring */ |
| supported = PJ_TRUE; |
| } else if (err == KErrAlreadyExists && retry_cnt < MAX_RETRY) { |
| /* Seems that the previous session is still arround, |
| * let's wait before retrying. |
| */ |
| enum { RETRY_WAIT_TIME = 3000 }; /* in msecs */ |
| snd_wait(RETRY_WAIT_TIME); |
| } else { |
| /* Seems that this format is not supported */ |
| retry_cnt = MAX_RETRY; |
| } |
| } |
| |
| if (supported) { |
| pjmedia_format ext_fmt; |
| |
| switch(i) { |
| case 0: /* AMRNB */ |
| pjmedia_format_init_audio(&ext_fmt, PJMEDIA_FORMAT_AMR, |
| 8000, 1, 16, 20, 7400, 12200); |
| af->dev_info.ext_fmt[fmt_cnt] = ext_fmt; |
| //af->dev_info.ext_fmt[fmt_cnt].vad = PJ_TRUE; |
| ++fmt_cnt; |
| break; |
| case 1: /* G.711 */ |
| pjmedia_format_init_audio(&ext_fmt, PJMEDIA_FORMAT_PCMU, |
| 8000, 1, 16, 20, 64000, 64000); |
| af->dev_info.ext_fmt[fmt_cnt] = ext_fmt; |
| //af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; |
| ++fmt_cnt; |
| pjmedia_format_init_audio(&ext_fmt, PJMEDIA_FORMAT_PCMA, |
| 8000, 1, 16, 20, 64000, 64000); |
| af->dev_info.ext_fmt[fmt_cnt] = ext_fmt; |
| //af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; |
| ++fmt_cnt; |
| g711_supported = PJ_TRUE; |
| break; |
| case 2: /* G.729 */ |
| pjmedia_format_init_audio(&ext_fmt, PJMEDIA_FORMAT_G729, |
| 8000, 1, 16, 20, 8000, 8000); |
| af->dev_info.ext_fmt[fmt_cnt] = ext_fmt; |
| //af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; |
| ++fmt_cnt; |
| break; |
| case 3: /* iLBC */ |
| pjmedia_format_init_audio(&ext_fmt, PJMEDIA_FORMAT_ILBC, |
| 8000, 1, 16, 30, 13333, 15200); |
| af->dev_info.ext_fmt[fmt_cnt] = ext_fmt; |
| //af->dev_info.ext_fmt[fmt_cnt].vad = PJ_TRUE; |
| ++fmt_cnt; |
| break; |
| } |
| } |
| } |
| |
| af->dev_info.ext_fmt_cnt = fmt_cnt; |
| |
| PJ_LOG(4, (THIS_FILE, "APS initialized")); |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: destroy factory */ |
| static pj_status_t factory_destroy(pjmedia_aud_dev_factory *f) |
| { |
| struct aps_factory *af = (struct aps_factory*)f; |
| pj_pool_t *pool = af->pool; |
| |
| af->pool = NULL; |
| pj_pool_release(pool); |
| |
| PJ_LOG(4, (THIS_FILE, "APS destroyed")); |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: refresh the device list */ |
| static pj_status_t factory_refresh(pjmedia_aud_dev_factory *f) |
| { |
| PJ_UNUSED_ARG(f); |
| return PJ_ENOTSUP; |
| } |
| |
| /* API: get number of devices */ |
| static unsigned factory_get_dev_count(pjmedia_aud_dev_factory *f) |
| { |
| PJ_UNUSED_ARG(f); |
| return 1; |
| } |
| |
| /* API: get device info */ |
| static pj_status_t factory_get_dev_info(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_dev_info *info) |
| { |
| struct aps_factory *af = (struct aps_factory*)f; |
| |
| PJ_ASSERT_RETURN(index == 0, PJMEDIA_EAUD_INVDEV); |
| |
| pj_memcpy(info, &af->dev_info, sizeof(*info)); |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: create default device parameter */ |
| static pj_status_t factory_default_param(pjmedia_aud_dev_factory *f, |
| unsigned index, |
| pjmedia_aud_param *param) |
| { |
| struct aps_factory *af = (struct aps_factory*)f; |
| |
| PJ_ASSERT_RETURN(index == 0, PJMEDIA_EAUD_INVDEV); |
| |
| pj_bzero(param, sizeof(*param)); |
| param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK; |
| param->rec_id = index; |
| param->play_id = index; |
| param->clock_rate = af->dev_info.default_samples_per_sec; |
| param->channel_count = 1; |
| param->samples_per_frame = af->dev_info.default_samples_per_sec * 20 / 1000; |
| param->bits_per_sample = BITS_PER_SAMPLE; |
| param->flags = PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE; |
| param->output_route = PJMEDIA_AUD_DEV_ROUTE_EARPIECE; |
| |
| return PJ_SUCCESS; |
| } |
| |
| |
| /* API: create stream */ |
| static pj_status_t factory_create_stream(pjmedia_aud_dev_factory *f, |
| const pjmedia_aud_param *param, |
| pjmedia_aud_rec_cb rec_cb, |
| pjmedia_aud_play_cb play_cb, |
| void *user_data, |
| pjmedia_aud_stream **p_aud_strm) |
| { |
| struct aps_factory *af = (struct aps_factory*)f; |
| pj_pool_t *pool; |
| struct aps_stream *strm; |
| |
| CPjAudioSetting aps_setting; |
| PjAudioCallback aps_rec_cb; |
| PjAudioCallback aps_play_cb; |
| |
| /* Can only support 16bits per sample */ |
| PJ_ASSERT_RETURN(param->bits_per_sample == BITS_PER_SAMPLE, PJ_EINVAL); |
| |
| /* Supported clock rates: |
| * - for non-PCM format: 8kHz |
| * - for PCM format: 8kHz and 16kHz |
| */ |
| PJ_ASSERT_RETURN(param->clock_rate == 8000 || |
| (param->clock_rate == 16000 && |
| param->ext_fmt.id == PJMEDIA_FORMAT_L16), |
| PJ_EINVAL); |
| |
| /* Supported channels number: |
| * - for non-PCM format: mono |
| * - for PCM format: mono and stereo |
| */ |
| PJ_ASSERT_RETURN(param->channel_count == 1 || |
| (param->channel_count == 2 && |
| param->ext_fmt.id == PJMEDIA_FORMAT_L16), |
| PJ_EINVAL); |
| |
| /* Create and Initialize stream descriptor */ |
| pool = pj_pool_create(af->pf, "aps-dev", 1000, 1000, NULL); |
| PJ_ASSERT_RETURN(pool, PJ_ENOMEM); |
| |
| strm = PJ_POOL_ZALLOC_T(pool, struct aps_stream); |
| strm->pool = pool; |
| strm->param = *param; |
| |
| if (strm->param.flags & PJMEDIA_AUD_DEV_CAP_EXT_FORMAT == 0) |
| strm->param.ext_fmt.id = PJMEDIA_FORMAT_L16; |
| |
| /* Set audio engine fourcc. */ |
| switch(strm->param.ext_fmt.id) { |
| case PJMEDIA_FORMAT_L16: |
| case PJMEDIA_FORMAT_PCMU: |
| case PJMEDIA_FORMAT_PCMA: |
| aps_setting.fourcc = TFourCC(KMCPFourCCIdG711); |
| break; |
| case PJMEDIA_FORMAT_AMR: |
| aps_setting.fourcc = TFourCC(KMCPFourCCIdAMRNB); |
| break; |
| case PJMEDIA_FORMAT_G729: |
| aps_setting.fourcc = TFourCC(KMCPFourCCIdG729); |
| break; |
| case PJMEDIA_FORMAT_ILBC: |
| aps_setting.fourcc = TFourCC(KMCPFourCCIdILBC); |
| break; |
| default: |
| aps_setting.fourcc = 0; |
| break; |
| } |
| |
| /* Set audio engine mode. */ |
| if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_AMR) |
| { |
| aps_setting.mode = (TAPSCodecMode)strm->param.ext_fmt.det.aud.avg_bps; |
| } |
| else if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMU || |
| strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 || |
| (strm->param.ext_fmt.id == PJMEDIA_FORMAT_ILBC && |
| strm->param.ext_fmt.det.aud.avg_bps != 15200)) |
| { |
| aps_setting.mode = EULawOr30ms; |
| } |
| else if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMA || |
| (strm->param.ext_fmt.id == PJMEDIA_FORMAT_ILBC && |
| strm->param.ext_fmt.det.aud.avg_bps == 15200)) |
| { |
| aps_setting.mode = EALawOr20ms; |
| } |
| |
| /* Disable VAD on L16, G711, and also G729 (G729's VAD potentially |
| * causes noise?). |
| */ |
| if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMU || |
| strm->param.ext_fmt.id == PJMEDIA_FORMAT_PCMA || |
| strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 || |
| strm->param.ext_fmt.id == PJMEDIA_FORMAT_G729) |
| { |
| aps_setting.vad = EFalse; |
| } else { |
| aps_setting.vad = (strm->param.flags & PJMEDIA_AUD_DEV_CAP_VAD) && |
| strm->param.vad_enabled; |
| } |
| |
| /* Set other audio engine attributes. */ |
| aps_setting.plc = (strm->param.flags & PJMEDIA_AUD_DEV_CAP_PLC) && |
| strm->param.plc_enabled; |
| aps_setting.cng = aps_setting.vad; |
| aps_setting.loudspk = |
| strm->param.output_route==PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER; |
| |
| /* Set audio engine callbacks. */ |
| if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16) { |
| aps_play_cb = &PlayCbPcm; |
| aps_rec_cb = &RecCbPcm; |
| } else { |
| aps_play_cb = &PlayCb; |
| aps_rec_cb = &RecCb; |
| } |
| |
| strm->rec_cb = rec_cb; |
| strm->play_cb = play_cb; |
| strm->user_data = user_data; |
| strm->resample_factor = strm->param.clock_rate / 8000; |
| |
| /* play_buf size is samples per frame scaled in to 8kHz mono. */ |
| strm->play_buf = (pj_int16_t*)pj_pool_zalloc( |
| pool, |
| (strm->param.samples_per_frame / |
| strm->resample_factor / |
| strm->param.channel_count) << 1); |
| strm->play_buf_len = 0; |
| strm->play_buf_start = 0; |
| |
| /* rec_buf size is samples per frame scaled in to 8kHz mono. */ |
| strm->rec_buf = (pj_int16_t*)pj_pool_zalloc( |
| pool, |
| (strm->param.samples_per_frame / |
| strm->resample_factor / |
| strm->param.channel_count) << 1); |
| strm->rec_buf_len = 0; |
| |
| if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_G729) { |
| TBitStream *g729_bitstream = new TBitStream; |
| |
| PJ_ASSERT_RETURN(g729_bitstream, PJ_ENOMEM); |
| strm->strm_data = (void*)g729_bitstream; |
| } |
| |
| /* Init resampler when format is PCM and clock rate is not 8kHz */ |
| if (strm->param.clock_rate != 8000 && |
| strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16) |
| { |
| pj_status_t status; |
| |
| if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { |
| /* Create resample for recorder */ |
| status = pjmedia_resample_create( pool, PJ_TRUE, PJ_FALSE, 1, |
| 8000, |
| strm->param.clock_rate, |
| 80, |
| &strm->rec_resample); |
| if (status != PJ_SUCCESS) |
| return status; |
| } |
| |
| if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { |
| /* Create resample for player */ |
| status = pjmedia_resample_create( pool, PJ_TRUE, PJ_FALSE, 1, |
| strm->param.clock_rate, |
| 8000, |
| 80 * strm->resample_factor, |
| &strm->play_resample); |
| if (status != PJ_SUCCESS) |
| return status; |
| } |
| } |
| |
| /* Create PCM buffer, when the clock rate is not 8kHz or not mono */ |
| if (strm->param.ext_fmt.id == PJMEDIA_FORMAT_L16 && |
| (strm->resample_factor > 1 || strm->param.channel_count != 1)) |
| { |
| strm->pcm_buf = (pj_int16_t*)pj_pool_zalloc(pool, |
| strm->param.samples_per_frame << 1); |
| } |
| |
| |
| /* Create the audio engine. */ |
| TRAPD(err, strm->engine = CPjAudioEngine::NewL(strm, |
| aps_rec_cb, aps_play_cb, |
| strm, aps_setting)); |
| if (err != KErrNone) { |
| pj_pool_release(pool); |
| return PJ_RETURN_OS_ERROR(err); |
| } |
| |
| /* Apply output volume setting if specified */ |
| if (param->flags & PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING) { |
| stream_set_cap(&strm->base, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, |
| ¶m->output_vol); |
| } |
| |
| /* Done */ |
| strm->base.op = &stream_op; |
| *p_aud_strm = &strm->base; |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: Get stream info. */ |
| static pj_status_t stream_get_param(pjmedia_aud_stream *s, |
| pjmedia_aud_param *pi) |
| { |
| struct aps_stream *strm = (struct aps_stream*)s; |
| |
| PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL); |
| |
| pj_memcpy(pi, &strm->param, sizeof(*pi)); |
| |
| /* Update the output volume setting */ |
| if (stream_get_cap(s, PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, |
| &pi->output_vol) == PJ_SUCCESS) |
| { |
| pi->flags |= PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; |
| } |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: get capability */ |
| static pj_status_t stream_get_cap(pjmedia_aud_stream *s, |
| pjmedia_aud_dev_cap cap, |
| void *pval) |
| { |
| struct aps_stream *strm = (struct aps_stream*)s; |
| pj_status_t status = PJ_ENOTSUP; |
| |
| PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); |
| |
| switch (cap) { |
| case PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE: |
| if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { |
| *(pjmedia_aud_dev_route*)pval = strm->param.output_route; |
| status = PJ_SUCCESS; |
| } |
| break; |
| |
| /* There is a case that GetMaxGain() stucks, e.g: in N95. */ |
| /* |
| case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING: |
| if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { |
| PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL); |
| |
| TInt max_gain = strm->engine->GetMaxGain(); |
| TInt gain = strm->engine->GetGain(); |
| |
| if (max_gain > 0 && gain >= 0) { |
| *(unsigned*)pval = gain * 100 / max_gain; |
| status = PJ_SUCCESS; |
| } else { |
| status = PJMEDIA_EAUD_NOTREADY; |
| } |
| } |
| break; |
| */ |
| |
| case PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING: |
| if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { |
| PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL); |
| |
| TInt max_vol = strm->engine->GetMaxVolume(); |
| TInt vol = strm->engine->GetVolume(); |
| |
| if (max_vol > 0 && vol >= 0) { |
| *(unsigned*)pval = vol * 100 / max_vol; |
| status = PJ_SUCCESS; |
| } else { |
| status = PJMEDIA_EAUD_NOTREADY; |
| } |
| } |
| break; |
| default: |
| break; |
| } |
| |
| return status; |
| } |
| |
| /* API: set capability */ |
| static pj_status_t stream_set_cap(pjmedia_aud_stream *s, |
| pjmedia_aud_dev_cap cap, |
| const void *pval) |
| { |
| struct aps_stream *strm = (struct aps_stream*)s; |
| pj_status_t status = PJ_ENOTSUP; |
| |
| PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); |
| |
| switch (cap) { |
| case PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE: |
| if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { |
| pjmedia_aud_dev_route r = *(const pjmedia_aud_dev_route*)pval; |
| TInt err; |
| |
| PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL); |
| |
| switch (r) { |
| case PJMEDIA_AUD_DEV_ROUTE_DEFAULT: |
| case PJMEDIA_AUD_DEV_ROUTE_EARPIECE: |
| err = strm->engine->ActivateSpeaker(EFalse); |
| status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err); |
| break; |
| case PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER: |
| err = strm->engine->ActivateSpeaker(ETrue); |
| status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err); |
| break; |
| default: |
| status = PJ_EINVAL; |
| break; |
| } |
| if (status == PJ_SUCCESS) |
| strm->param.output_route = r; |
| } |
| break; |
| |
| /* There is a case that GetMaxGain() stucks, e.g: in N95. */ |
| /* |
| case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING: |
| if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { |
| PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL); |
| |
| TInt max_gain = strm->engine->GetMaxGain(); |
| if (max_gain > 0) { |
| TInt gain, err; |
| |
| gain = *(unsigned*)pval * max_gain / 100; |
| err = strm->engine->SetGain(gain); |
| status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err); |
| } else { |
| status = PJMEDIA_EAUD_NOTREADY; |
| } |
| if (status == PJ_SUCCESS) |
| strm->param.input_vol = *(unsigned*)pval; |
| } |
| break; |
| */ |
| |
| case PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING: |
| if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { |
| PJ_ASSERT_RETURN(strm->engine, PJ_EINVAL); |
| |
| TInt max_vol = strm->engine->GetMaxVolume(); |
| if (max_vol > 0) { |
| TInt vol, err; |
| |
| vol = *(unsigned*)pval * max_vol / 100; |
| err = strm->engine->SetVolume(vol); |
| status = (err==KErrNone)? PJ_SUCCESS:PJ_RETURN_OS_ERROR(err); |
| } else { |
| status = PJMEDIA_EAUD_NOTREADY; |
| } |
| if (status == PJ_SUCCESS) |
| strm->param.output_vol = *(unsigned*)pval; |
| } |
| break; |
| default: |
| break; |
| } |
| |
| return status; |
| } |
| |
| /* API: Start stream. */ |
| static pj_status_t stream_start(pjmedia_aud_stream *strm) |
| { |
| struct aps_stream *stream = (struct aps_stream*)strm; |
| |
| PJ_ASSERT_RETURN(stream, PJ_EINVAL); |
| |
| if (stream->engine) { |
| TInt err = stream->engine->StartL(); |
| if (err != KErrNone) |
| return PJ_RETURN_OS_ERROR(err); |
| } |
| |
| return PJ_SUCCESS; |
| } |
| |
| /* API: Stop stream. */ |
| static pj_status_t stream_stop(pjmedia_aud_stream *strm) |
| { |
| struct aps_stream *stream = (struct aps_stream*)strm; |
| |
| PJ_ASSERT_RETURN(stream, PJ_EINVAL); |
| |
| if (stream->engine) { |
| stream->engine->Stop(); |
| } |
| |
| return PJ_SUCCESS; |
| } |
| |
| |
| /* API: Destroy stream. */ |
| static pj_status_t stream_destroy(pjmedia_aud_stream *strm) |
| { |
| struct aps_stream *stream = (struct aps_stream*)strm; |
| |
| PJ_ASSERT_RETURN(stream, PJ_EINVAL); |
| |
| stream_stop(strm); |
| |
| delete stream->engine; |
| stream->engine = NULL; |
| |
| if (stream->param.ext_fmt.id == PJMEDIA_FORMAT_G729) { |
| TBitStream *g729_bitstream = (TBitStream*)stream->strm_data; |
| stream->strm_data = NULL; |
| delete g729_bitstream; |
| } |
| |
| pj_pool_t *pool; |
| pool = stream->pool; |
| if (pool) { |
| stream->pool = NULL; |
| pj_pool_release(pool); |
| } |
| |
| return PJ_SUCCESS; |
| } |
| |
| #endif // PJMEDIA_AUDIO_DEV_HAS_SYMB_APS |
| |