From 1dacdee696b7591a6dcc0b3c1d0f41573e473168 Mon Sep 17 00:00:00 2001 From: Benny Prijono Date: Thu, 12 Mar 2009 18:11:37 +0000 Subject: (Major) Task #737 and #738: integration of APS-Direct and Audiodev from aps-direct branch to trunk. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2506 74dad513-b988-da41-8d7b-12977e46ad98 --- pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp | 1110 +++++++++++++++++++++++++ 1 file changed, 1110 insertions(+) create mode 100644 pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp (limited to 'pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp') diff --git a/pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp b/pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp new file mode 100644 index 00000000..f9437e55 --- /dev/null +++ b/pjmedia/src/pjmedia-audiodev/symb_mda_dev.cpp @@ -0,0 +1,1110 @@ +/* $Id$ */ +/* + * Copyright (C) 2008-2009 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 + +#if PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA + +/* + * This file provides sound implementation for Symbian Audio Streaming + * device. Application using this sound abstraction must link with: + * - mediaclientaudiostream.lib, and + * - mediaclientaudioinputstream.lib + */ +#include +#include +#include + + +#define THIS_FILE "symb_mda_dev.c" +#define BITS_PER_SAMPLE 16 +#define BYTES_PER_SAMPLE (BITS_PER_SAMPLE/8) + + +#if 1 +# define TRACE_(st) PJ_LOG(3, st) +#else +# define TRACE_(st) +#endif + + +/* MDA factory */ +struct mda_factory +{ + pjmedia_aud_dev_factory base; + pj_pool_t *pool; + pj_pool_factory *pf; + pjmedia_aud_dev_info dev_info; +}; + +/* Forward declaration of internal engine. */ +class CPjAudioInputEngine; +class CPjAudioOutputEngine; + +/* MDA stream. */ +struct mda_stream +{ + // Base + pjmedia_aud_stream base; /**< Base class. */ + + // Pool + pj_pool_t *pool; /**< Memory pool. */ + + // Common settings. + pjmedia_aud_param param; /**< Stream param. */ + + // Audio engine + CPjAudioInputEngine *in_engine; /**< Record engine. */ + CPjAudioOutputEngine *out_engine; /**< Playback engine. */ +}; + + +/* Prototypes */ +static pj_status_t factory_init(pjmedia_aud_dev_factory *f); +static pj_status_t factory_destroy(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 +}; + +static pjmedia_aud_stream_op stream_op = +{ + &stream_get_param, + &stream_get_cap, + &stream_set_cap, + &stream_start, + &stream_stop, + &stream_destroy +}; + + +/* + * Convert clock rate to Symbian's TMdaAudioDataSettings capability. + */ +static TInt get_clock_rate_cap(unsigned clock_rate) +{ + switch (clock_rate) { + case 8000: return TMdaAudioDataSettings::ESampleRate8000Hz; + case 11025: return TMdaAudioDataSettings::ESampleRate11025Hz; + case 12000: return TMdaAudioDataSettings::ESampleRate12000Hz; + case 16000: return TMdaAudioDataSettings::ESampleRate16000Hz; + case 22050: return TMdaAudioDataSettings::ESampleRate22050Hz; + case 24000: return TMdaAudioDataSettings::ESampleRate24000Hz; + case 32000: return TMdaAudioDataSettings::ESampleRate32000Hz; + case 44100: return TMdaAudioDataSettings::ESampleRate44100Hz; + case 48000: return TMdaAudioDataSettings::ESampleRate48000Hz; + case 64000: return TMdaAudioDataSettings::ESampleRate64000Hz; + case 96000: return TMdaAudioDataSettings::ESampleRate96000Hz; + default: + return 0; + } +} + +/* + * Convert number of channels into Symbian's TMdaAudioDataSettings capability. + */ +static TInt get_channel_cap(unsigned channel_count) +{ + switch (channel_count) { + case 1: return TMdaAudioDataSettings::EChannelsMono; + case 2: return TMdaAudioDataSettings::EChannelsStereo; + default: + return 0; + } +} + +/* + * 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)); +} + +////////////////////////////////////////////////////////////////////////////// +// + +/* + * Implementation: Symbian Input Stream. + */ +class CPjAudioInputEngine : public CBase, MMdaAudioInputStreamCallback +{ +public: + enum State + { + STATE_INACTIVE, + STATE_ACTIVE, + }; + + ~CPjAudioInputEngine(); + + static CPjAudioInputEngine *NewL(struct mda_stream *parent_strm, + pjmedia_aud_rec_cb rec_cb, + void *user_data); + + static CPjAudioInputEngine *NewLC(struct mda_stream *parent_strm, + pjmedia_aud_rec_cb rec_cb, + void *user_data); + + pj_status_t StartRecord(); + void Stop(); + + pj_status_t SetGain(TInt gain) { + if (iInputStream_) { + iInputStream_->SetGain(gain); + return PJ_SUCCESS; + } else + return PJ_EINVALIDOP; + } + + TInt GetGain() { + if (iInputStream_) { + return iInputStream_->Gain(); + } else + return PJ_EINVALIDOP; + } + + TInt GetMaxGain() { + if (iInputStream_) { + return iInputStream_->MaxGain(); + } else + return PJ_EINVALIDOP; + } + +private: + State state_; + struct mda_stream *parentStrm_; + pjmedia_aud_rec_cb recCb_; + void *userData_; + CMdaAudioInputStream *iInputStream_; + HBufC8 *iStreamBuffer_; + TPtr8 iFramePtr_; + TInt lastError_; + pj_uint32_t timeStamp_; + + // cache variable + // to avoid calculating frame length repeatedly + TInt frameLen_; + + // sometimes recorded size != requested framesize, so let's + // provide a buffer to make sure the rec callback returning + // framesize as requested. + TUint8 *frameRecBuf_; + TInt frameRecBufLen_; + + CPjAudioInputEngine(struct mda_stream *parent_strm, + pjmedia_aud_rec_cb rec_cb, + void *user_data); + void ConstructL(); + TPtr8 & GetFrame(); + +public: + virtual void MaiscOpenComplete(TInt aError); + virtual void MaiscBufferCopied(TInt aError, const TDesC8 &aBuffer); + virtual void MaiscRecordComplete(TInt aError); + +}; + + +CPjAudioInputEngine::CPjAudioInputEngine(struct mda_stream *parent_strm, + pjmedia_aud_rec_cb rec_cb, + void *user_data) + : state_(STATE_INACTIVE), parentStrm_(parent_strm), + recCb_(rec_cb), userData_(user_data), + iInputStream_(NULL), iStreamBuffer_(NULL), iFramePtr_(0, 0), + lastError_(KErrNone), timeStamp_(0), + frameLen_(parent_strm->param.samples_per_frame * + parent_strm->param.channel_count * + BYTES_PER_SAMPLE), + frameRecBuf_(NULL), frameRecBufLen_(0) +{ +} + +CPjAudioInputEngine::~CPjAudioInputEngine() +{ + Stop(); + + delete iStreamBuffer_; + iStreamBuffer_ = NULL; + + delete [] frameRecBuf_; + frameRecBuf_ = NULL; + frameRecBufLen_ = 0; +} + +void CPjAudioInputEngine::ConstructL() +{ + iStreamBuffer_ = HBufC8::NewL(frameLen_); + CleanupStack::PushL(iStreamBuffer_); + + frameRecBuf_ = new TUint8[frameLen_*2]; + CleanupStack::PushL(frameRecBuf_); +} + +CPjAudioInputEngine *CPjAudioInputEngine::NewLC(struct mda_stream *parent, + pjmedia_aud_rec_cb rec_cb, + void *user_data) +{ + CPjAudioInputEngine* self = new (ELeave) CPjAudioInputEngine(parent, + rec_cb, + user_data); + CleanupStack::PushL(self); + self->ConstructL(); + return self; +} + +CPjAudioInputEngine *CPjAudioInputEngine::NewL(struct mda_stream *parent, + pjmedia_aud_rec_cb rec_cb, + void *user_data) +{ + CPjAudioInputEngine *self = NewLC(parent, rec_cb, user_data); + CleanupStack::Pop(self->frameRecBuf_); + CleanupStack::Pop(self->iStreamBuffer_); + CleanupStack::Pop(self); + return self; +} + + +pj_status_t CPjAudioInputEngine::StartRecord() +{ + + // Ignore command if recording is in progress. + if (state_ == STATE_ACTIVE) + return PJ_SUCCESS; + + // According to Nokia's AudioStream example, some 2nd Edition, FP2 devices + // (such as Nokia 6630) require the stream to be reconstructed each time + // before calling Open() - otherwise the callback never gets called. + // For uniform behavior, lets just delete/re-create the stream for all + // devices. + + // Destroy existing stream. + if (iInputStream_) delete iInputStream_; + iInputStream_ = NULL; + + // Create the stream. + TRAPD(err, iInputStream_ = CMdaAudioInputStream::NewL(*this)); + if (err != KErrNone) + return PJ_RETURN_OS_ERROR(err); + + // Initialize settings. + TMdaAudioDataSettings iStreamSettings; + iStreamSettings.iChannels = + get_channel_cap(parentStrm_->param.channel_count); + iStreamSettings.iSampleRate = + get_clock_rate_cap(parentStrm_->param.clock_rate); + + pj_assert(iStreamSettings.iChannels != 0 && + iStreamSettings.iSampleRate != 0); + + PJ_LOG(4,(THIS_FILE, "Opening sound device for capture, " + "clock rate=%d, channel count=%d..", + parentStrm_->param.clock_rate, + parentStrm_->param.channel_count)); + + // Open stream. + lastError_ = KRequestPending; + iInputStream_->Open(&iStreamSettings); + + // Success + PJ_LOG(4,(THIS_FILE, "Sound capture started.")); + return PJ_SUCCESS; +} + + +void CPjAudioInputEngine::Stop() +{ + // If capture is in progress, stop it. + if (iInputStream_ && state_ == STATE_ACTIVE) { + lastError_ = KRequestPending; + iInputStream_->Stop(); + + // Wait until it's actually stopped + while (lastError_ == KRequestPending) + pj_symbianos_poll(-1, 100); + } + + if (iInputStream_) { + delete iInputStream_; + iInputStream_ = NULL; + } + + state_ = STATE_INACTIVE; +} + + +TPtr8 & CPjAudioInputEngine::GetFrame() +{ + //iStreamBuffer_->Des().FillZ(frameLen_); + iFramePtr_.Set((TUint8*)(iStreamBuffer_->Ptr()), frameLen_, frameLen_); + return iFramePtr_; +} + +void CPjAudioInputEngine::MaiscOpenComplete(TInt aError) +{ + lastError_ = aError; + if (aError != KErrNone) { + snd_perror("Error in MaiscOpenComplete()", aError); + return; + } + + // set stream priority to normal and time sensitive + iInputStream_->SetPriority(EPriorityNormal, + EMdaPriorityPreferenceTime); + + // Read the first frame. + TPtr8 & frm = GetFrame(); + TRAPD(err2, iInputStream_->ReadL(frm)); + if (err2) { + PJ_LOG(4,(THIS_FILE, "Exception in iInputStream_->ReadL()")); + } +} + +void CPjAudioInputEngine::MaiscBufferCopied(TInt aError, + const TDesC8 &aBuffer) +{ + lastError_ = aError; + if (aError != KErrNone) { + snd_perror("Error in MaiscBufferCopied()", aError); + return; + } + + if (frameRecBufLen_ || aBuffer.Size() < frameLen_) { + pj_memcpy(frameRecBuf_ + frameRecBufLen_, (void*) aBuffer.Ptr(), aBuffer.Size()); + frameRecBufLen_ += aBuffer.Size(); + } + + if (frameRecBufLen_) { + while (frameRecBufLen_ >= frameLen_) { + pjmedia_frame f; + + f.type = PJMEDIA_FRAME_TYPE_AUDIO; + f.buf = frameRecBuf_; + f.size = frameLen_; + f.timestamp.u32.lo = timeStamp_; + f.bit_info = 0; + + // Call the callback. + recCb_(userData_, &f); + // Increment timestamp. + timeStamp_ += parentStrm_->param.samples_per_frame; + + frameRecBufLen_ -= frameLen_; + pj_memmove(frameRecBuf_, frameRecBuf_+frameLen_, frameRecBufLen_); + } + } else { + pjmedia_frame f; + + f.type = PJMEDIA_FRAME_TYPE_AUDIO; + f.buf = (void*)aBuffer.Ptr(); + f.size = aBuffer.Size(); + f.timestamp.u32.lo = timeStamp_; + f.bit_info = 0; + + // Call the callback. + recCb_(userData_, &f); + + // Increment timestamp. + timeStamp_ += parentStrm_->param.samples_per_frame; + } + + // Record next frame + TPtr8 & frm = GetFrame(); + TRAPD(err2, iInputStream_->ReadL(frm)); + if (err2) { + PJ_LOG(4,(THIS_FILE, "Exception in iInputStream_->ReadL()")); + } +} + + +void CPjAudioInputEngine::MaiscRecordComplete(TInt aError) +{ + lastError_ = aError; + state_ = STATE_INACTIVE; + if (aError != KErrNone) { + snd_perror("Error in MaiscRecordComplete()", aError); + } +} + + + +////////////////////////////////////////////////////////////////////////////// +// + +/* + * Implementation: Symbian Output Stream. + */ + +class CPjAudioOutputEngine : public CBase, MMdaAudioOutputStreamCallback +{ +public: + enum State + { + STATE_INACTIVE, + STATE_ACTIVE, + }; + + ~CPjAudioOutputEngine(); + + static CPjAudioOutputEngine *NewL(struct mda_stream *parent_strm, + pjmedia_aud_play_cb play_cb, + void *user_data); + + static CPjAudioOutputEngine *NewLC(struct mda_stream *parent_strm, + pjmedia_aud_play_cb rec_cb, + void *user_data); + + pj_status_t StartPlay(); + void Stop(); + + pj_status_t SetVolume(TInt vol) { + if (iOutputStream_) { + iOutputStream_->SetVolume(vol); + return PJ_SUCCESS; + } else + return PJ_EINVALIDOP; + } + + TInt GetVolume() { + if (iOutputStream_) { + return iOutputStream_->Volume(); + } else + return PJ_EINVALIDOP; + } + + TInt GetMaxVolume() { + if (iOutputStream_) { + return iOutputStream_->MaxVolume(); + } else + return PJ_EINVALIDOP; + } + +private: + State state_; + struct mda_stream *parentStrm_; + pjmedia_aud_play_cb playCb_; + void *userData_; + CMdaAudioOutputStream *iOutputStream_; + TUint8 *frameBuf_; + unsigned frameBufSize_; + TPtrC8 frame_; + TInt lastError_; + unsigned timestamp_; + + CPjAudioOutputEngine(struct mda_stream *parent_strm, + pjmedia_aud_play_cb play_cb, + void *user_data); + void ConstructL(); + + virtual void MaoscOpenComplete(TInt aError); + virtual void MaoscBufferCopied(TInt aError, const TDesC8& aBuffer); + virtual void MaoscPlayComplete(TInt aError); +}; + + +CPjAudioOutputEngine::CPjAudioOutputEngine(struct mda_stream *parent_strm, + pjmedia_aud_play_cb play_cb, + void *user_data) +: state_(STATE_INACTIVE), parentStrm_(parent_strm), playCb_(play_cb), + userData_(user_data), iOutputStream_(NULL), frameBuf_(NULL), + lastError_(KErrNone), timestamp_(0) +{ +} + + +void CPjAudioOutputEngine::ConstructL() +{ + frameBufSize_ = parentStrm_->param.samples_per_frame * + parentStrm_->param.channel_count * + BYTES_PER_SAMPLE; + frameBuf_ = new TUint8[frameBufSize_]; +} + +CPjAudioOutputEngine::~CPjAudioOutputEngine() +{ + Stop(); + delete [] frameBuf_; +} + +CPjAudioOutputEngine * +CPjAudioOutputEngine::NewLC(struct mda_stream *parent_strm, + pjmedia_aud_play_cb play_cb, + void *user_data) +{ + CPjAudioOutputEngine* self = new (ELeave) CPjAudioOutputEngine(parent_strm, + play_cb, + user_data); + CleanupStack::PushL(self); + self->ConstructL(); + return self; +} + +CPjAudioOutputEngine * +CPjAudioOutputEngine::NewL(struct mda_stream *parent_strm, + pjmedia_aud_play_cb play_cb, + void *user_data) +{ + CPjAudioOutputEngine *self = NewLC(parent_strm, play_cb, user_data); + CleanupStack::Pop(self); + return self; +} + +pj_status_t CPjAudioOutputEngine::StartPlay() +{ + // Ignore command if playing is in progress. + if (state_ == STATE_ACTIVE) + return PJ_SUCCESS; + + // Destroy existing stream. + if (iOutputStream_) delete iOutputStream_; + iOutputStream_ = NULL; + + // Create the stream + TRAPD(err, iOutputStream_ = CMdaAudioOutputStream::NewL(*this)); + if (err != KErrNone) + return PJ_RETURN_OS_ERROR(err); + + // Initialize settings. + TMdaAudioDataSettings iStreamSettings; + iStreamSettings.iChannels = + get_channel_cap(parentStrm_->param.channel_count); + iStreamSettings.iSampleRate = + get_clock_rate_cap(parentStrm_->param.clock_rate); + + pj_assert(iStreamSettings.iChannels != 0 && + iStreamSettings.iSampleRate != 0); + + PJ_LOG(4,(THIS_FILE, "Opening sound device for playback, " + "clock rate=%d, channel count=%d..", + parentStrm_->param.clock_rate, + parentStrm_->param.channel_count)); + + // Open stream. + lastError_ = KRequestPending; + iOutputStream_->Open(&iStreamSettings); + + // Success + PJ_LOG(4,(THIS_FILE, "Sound playback started")); + return PJ_SUCCESS; + +} + +void CPjAudioOutputEngine::Stop() +{ + // Stop stream if it's playing + if (iOutputStream_ && state_ != STATE_INACTIVE) { + lastError_ = KRequestPending; + iOutputStream_->Stop(); + + // Wait until it's actually stopped + while (lastError_ == KRequestPending) + pj_symbianos_poll(-1, 100); + } + + if (iOutputStream_) { + delete iOutputStream_; + iOutputStream_ = NULL; + } + + state_ = STATE_INACTIVE; +} + +void CPjAudioOutputEngine::MaoscOpenComplete(TInt aError) +{ + lastError_ = aError; + + if (aError==KErrNone) { + // output stream opened succesfully, set status to Active + state_ = STATE_ACTIVE; + + // set stream properties, 16bit 8KHz mono + TMdaAudioDataSettings iSettings; + iSettings.iChannels = + get_channel_cap(parentStrm_->param.channel_count); + iSettings.iSampleRate = + get_clock_rate_cap(parentStrm_->param.clock_rate); + + iOutputStream_->SetAudioPropertiesL(iSettings.iSampleRate, + iSettings.iChannels); + + // set volume to 1/2th of stream max volume + iOutputStream_->SetVolume(iOutputStream_->MaxVolume()/2); + + // set stream priority to normal and time sensitive + iOutputStream_->SetPriority(EPriorityNormal, + EMdaPriorityPreferenceTime); + + // Call callback to retrieve frame from upstream. + pjmedia_frame f; + pj_status_t status; + + f.type = PJMEDIA_FRAME_TYPE_AUDIO; + f.buf = frameBuf_; + f.size = frameBufSize_; + f.timestamp.u32.lo = timestamp_; + f.bit_info = 0; + + status = playCb_(this->userData_, &f); + if (status != PJ_SUCCESS) { + this->Stop(); + return; + } + + // Increment timestamp. + timestamp_ += (frameBufSize_ / BYTES_PER_SAMPLE); + + // issue WriteL() to write the first audio data block, + // subsequent calls to WriteL() will be issued in + // MMdaAudioOutputStreamCallback::MaoscBufferCopied() + // until whole data buffer is written. + frame_.Set(frameBuf_, frameBufSize_); + iOutputStream_->WriteL(frame_); + } else { + snd_perror("Error in MaoscOpenComplete()", aError); + } +} + +void CPjAudioOutputEngine::MaoscBufferCopied(TInt aError, + const TDesC8& aBuffer) +{ + PJ_UNUSED_ARG(aBuffer); + + if (aError==KErrNone) { + // Buffer successfully written, feed another one. + + // Call callback to retrieve frame from upstream. + pjmedia_frame f; + pj_status_t status; + + f.type = PJMEDIA_FRAME_TYPE_AUDIO; + f.buf = frameBuf_; + f.size = frameBufSize_; + f.timestamp.u32.lo = timestamp_; + f.bit_info = 0; + + status = playCb_(this->userData_, &f); + if (status != PJ_SUCCESS) { + this->Stop(); + return; + } + + // Increment timestamp. + timestamp_ += (frameBufSize_ / BYTES_PER_SAMPLE); + + // Write to playback stream. + frame_.Set(frameBuf_, frameBufSize_); + iOutputStream_->WriteL(frame_); + + } else if (aError==KErrAbort) { + // playing was aborted, due to call to CMdaAudioOutputStream::Stop() + state_ = STATE_INACTIVE; + } else { + // error writing data to output + lastError_ = aError; + state_ = STATE_INACTIVE; + snd_perror("Error in MaoscBufferCopied()", aError); + } +} + +void CPjAudioOutputEngine::MaoscPlayComplete(TInt aError) +{ + lastError_ = aError; + state_ = STATE_INACTIVE; + if (aError != KErrNone) { + snd_perror("Error in MaoscPlayComplete()", aError); + } +} + +/**************************************************************************** + * Factory operations + */ + +/* + * C compatible declaration of MDA factory. + */ +PJ_BEGIN_DECL +PJ_DECL(pjmedia_aud_dev_factory*) pjmedia_symb_mda_factory(pj_pool_factory *pf); +PJ_END_DECL + +/* + * Init Symbian audio driver. + */ +pjmedia_aud_dev_factory* pjmedia_symb_mda_factory(pj_pool_factory *pf) +{ + struct mda_factory *f; + pj_pool_t *pool; + + pool = pj_pool_create(pf, "symb_aud", 1000, 1000, NULL); + f = PJ_POOL_ZALLOC_T(pool, struct mda_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 mda_factory *af = (struct mda_factory*)f; + + pj_ansi_strcpy(af->dev_info.name, "Symbian Audio"); + af->dev_info.default_samples_per_sec = 8000; + af->dev_info.caps = PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING | + PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING; + af->dev_info.input_count = 1; + af->dev_info.output_count = 1; + + PJ_LOG(4, (THIS_FILE, "Symb Mda initialized")); + + return PJ_SUCCESS; +} + +/* API: destroy factory */ +static pj_status_t factory_destroy(pjmedia_aud_dev_factory *f) +{ + struct mda_factory *af = (struct mda_factory*)f; + pj_pool_t *pool = af->pool; + + af->pool = NULL; + pj_pool_release(pool); + + PJ_LOG(4, (THIS_FILE, "Symbian Mda destroyed")); + + return PJ_SUCCESS; +} + +/* 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 mda_factory *af = (struct mda_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 mda_factory *af = (struct mda_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 = af->dev_info.caps; + + 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 mda_factory *mf = (struct mda_factory*)f; + pj_pool_t *pool; + struct mda_stream *strm; + + /* Can only support 16bits per sample raw PCM format. */ + PJ_ASSERT_RETURN(param->bits_per_sample == BITS_PER_SAMPLE, PJ_EINVAL); + PJ_ASSERT_RETURN((param->flags & PJMEDIA_AUD_DEV_CAP_EXT_FORMAT)==0 || + param->ext_fmt.id == PJMEDIA_FORMAT_L16, + PJ_ENOTSUP); + + /* Create and Initialize stream descriptor */ + pool = pj_pool_create(mf->pf, "symb_aud_dev", 1000, 1000, NULL); + PJ_ASSERT_RETURN(pool, PJ_ENOMEM); + + strm = PJ_POOL_ZALLOC_T(pool, struct mda_stream); + strm->pool = pool; + strm->param = *param; + + // Create the output stream. + if (strm->param.dir & PJMEDIA_DIR_PLAYBACK) { + TRAPD(err, strm->out_engine = CPjAudioOutputEngine::NewL(strm, play_cb, + user_data)); + if (err != KErrNone) { + pj_pool_release(pool); + return PJ_RETURN_OS_ERROR(err); + } + } + + // Create the input stream. + if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { + TRAPD(err, strm->in_engine = CPjAudioInputEngine::NewL(strm, rec_cb, + user_data)); + if (err != KErrNone) { + strm->in_engine = NULL; + delete strm->out_engine; + strm->out_engine = NULL; + pj_pool_release(pool); + return PJ_RETURN_OS_ERROR(err); + } + } + + /* 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 mda_stream *strm = (struct mda_stream*)s; + + PJ_ASSERT_RETURN(strm && pi, PJ_EINVAL); + + pj_memcpy(pi, &strm->param, sizeof(*pi)); + + 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 mda_stream *strm = (struct mda_stream*)s; + pj_status_t status = PJ_ENOTSUP; + + PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); + + switch (cap) { + case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING: + if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { + PJ_ASSERT_RETURN(strm->in_engine, PJ_EINVAL); + + TInt max_gain = strm->in_engine->GetMaxGain(); + TInt gain = strm->in_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->out_engine, PJ_EINVAL); + + TInt max_vol = strm->out_engine->GetMaxVolume(); + TInt vol = strm->out_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 mda_stream *strm = (struct mda_stream*)s; + pj_status_t status = PJ_ENOTSUP; + + PJ_ASSERT_RETURN(s && pval, PJ_EINVAL); + + switch (cap) { + case PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING: + if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { + PJ_ASSERT_RETURN(strm->in_engine, PJ_EINVAL); + + TInt max_gain = strm->in_engine->GetMaxGain(); + if (max_gain > 0) { + TInt gain; + + gain = *(unsigned*)pval * max_gain / 100; + status = strm->in_engine->SetGain(gain); + } else { + status = PJMEDIA_EAUD_NOTREADY; + } + } + break; + case PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING: + if (strm->param.dir & PJMEDIA_DIR_CAPTURE) { + PJ_ASSERT_RETURN(strm->out_engine, PJ_EINVAL); + + TInt max_vol = strm->out_engine->GetMaxVolume(); + if (max_vol > 0) { + TInt vol; + + vol = *(unsigned*)pval * max_vol / 100; + status = strm->out_engine->SetVolume(vol); + } else { + status = PJMEDIA_EAUD_NOTREADY; + } + } + break; + default: + break; + } + + return status; +} + +/* API: Start stream. */ +static pj_status_t stream_start(pjmedia_aud_stream *strm) +{ + struct mda_stream *stream = (struct mda_stream*)strm; + + PJ_ASSERT_RETURN(stream, PJ_EINVAL); + + if (stream->out_engine) { + pj_status_t status; + status = stream->out_engine->StartPlay(); + if (status != PJ_SUCCESS) + return status; + } + + if (stream->in_engine) { + pj_status_t status; + status = stream->in_engine->StartRecord(); + if (status != PJ_SUCCESS) + return status; + } + + return PJ_SUCCESS; +} + +/* API: Stop stream. */ +static pj_status_t stream_stop(pjmedia_aud_stream *strm) +{ + struct mda_stream *stream = (struct mda_stream*)strm; + + PJ_ASSERT_RETURN(stream, PJ_EINVAL); + + if (stream->in_engine) { + stream->in_engine->Stop(); + } + + if (stream->out_engine) { + stream->out_engine->Stop(); + } + + return PJ_SUCCESS; +} + + +/* API: Destroy stream. */ +static pj_status_t stream_destroy(pjmedia_aud_stream *strm) +{ + struct mda_stream *stream = (struct mda_stream*)strm; + + PJ_ASSERT_RETURN(stream, PJ_EINVAL); + + stream_stop(strm); + + delete stream->in_engine; + stream->in_engine = NULL; + + delete stream->out_engine; + stream->out_engine = NULL; + + 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_MDA */ -- cgit v1.2.3