From f3ab456a17af1c89a6e3be4d20c5944853df1cb0 Mon Sep 17 00:00:00 2001 From: "David M. Lee" Date: Mon, 7 Jan 2013 14:24:28 -0600 Subject: Import pjproject-2.0.1 --- pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp | 1929 +++++++++++++++++++++++++ 1 file changed, 1929 insertions(+) create mode 100644 pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp (limited to 'pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp') diff --git a/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp b/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp new file mode 100644 index 0000000..d2c6564 --- /dev/null +++ b/pjmedia/src/pjmedia-audiodev/symb_aps_dev.cpp @@ -0,0 +1,1929 @@ +/* $Id: symb_aps_dev.cpp 3841 2011-10-24 09:28:13Z ming $ */ +/* + * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) + * Copyright (C) 2003-2008 Benny Prijono + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#if PJMEDIA_AUDIO_DEV_HAS_SYMB_APS + +#include +#include +#include +#include + +/* 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* aQ, + RMsgQueue* 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* aQ, + RMsgQueue* 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 *iQ; // (not owned) + RMsgQueue *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 iReadQ; + RMsgQueue iReadCommQ; + TBool readq_opened; + RMsgQueue iWriteQ; + RMsgQueue 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.bitrate == 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.bitrate == 15200 && + samples_cnt == 160) || + (strm->param.ext_fmt.bitrate != 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) { + switch(i) { + case 0: /* AMRNB */ + af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_AMR; + af->dev_info.ext_fmt[fmt_cnt].bitrate = 7400; + af->dev_info.ext_fmt[fmt_cnt].vad = PJ_TRUE; + ++fmt_cnt; + break; + case 1: /* G.711 */ + af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_PCMU; + af->dev_info.ext_fmt[fmt_cnt].bitrate = 64000; + af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; + ++fmt_cnt; + af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_PCMA; + af->dev_info.ext_fmt[fmt_cnt].bitrate = 64000; + af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; + ++fmt_cnt; + g711_supported = PJ_TRUE; + break; + case 2: /* G.729 */ + af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_G729; + af->dev_info.ext_fmt[fmt_cnt].bitrate = 8000; + af->dev_info.ext_fmt[fmt_cnt].vad = PJ_FALSE; + ++fmt_cnt; + break; + case 3: /* iLBC */ + af->dev_info.ext_fmt[fmt_cnt].id = PJMEDIA_FORMAT_ILBC; + af->dev_info.ext_fmt[fmt_cnt].bitrate = 13333; + 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.bitrate; + } + 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.bitrate != 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.bitrate == 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.ext_fmt.vad; + } + + /* Set other audio engine attributes. */ + aps_setting.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 + -- cgit v1.2.3