| /* |
| ** Copyright 2010, The Android Open-Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| #include <math.h> |
| |
| //#define LOG_NDEBUG 0 |
| |
| #define LOG_TAG "AudioHardware" |
| |
| #include <utils/Log.h> |
| #include <utils/String8.h> |
| |
| #include <stdio.h> |
| #include <unistd.h> |
| #include <sys/ioctl.h> |
| #include <sys/types.h> |
| #include <sys/stat.h> |
| #include <sys/resource.h> |
| #include <dlfcn.h> |
| #include <fcntl.h> |
| |
| #include "AudioHardware.h" |
| #include <media/AudioRecord.h> |
| #include <hardware_legacy/power.h> |
| |
| extern "C" { |
| #include "alsa_audio.h" |
| } |
| |
| |
| namespace android { |
| |
| const uint32_t AudioHardware::inputSamplingRates[] = { |
| 8000, 11025, 16000, 22050, 44100 |
| }; |
| |
| // trace driver operations for dump |
| // |
| #define DRIVER_TRACE |
| |
| enum { |
| DRV_NONE, |
| DRV_PCM_OPEN, |
| DRV_PCM_CLOSE, |
| DRV_PCM_WRITE, |
| DRV_PCM_READ, |
| DRV_MIXER_OPEN, |
| DRV_MIXER_CLOSE, |
| DRV_MIXER_GET, |
| DRV_MIXER_SEL |
| }; |
| |
| #ifdef DRIVER_TRACE |
| #define TRACE_DRIVER_IN(op) mDriverOp = op; |
| #define TRACE_DRIVER_OUT mDriverOp = DRV_NONE; |
| #else |
| #define TRACE_DRIVER_IN(op) |
| #define TRACE_DRIVER_OUT |
| #endif |
| |
| // ---------------------------------------------------------------------------- |
| |
| const char *AudioHardware::inputPathNameDefault = "Default"; |
| const char *AudioHardware::inputPathNameCamcorder = "Camcorder"; |
| const char *AudioHardware::inputPathNameVoiceRecognition = "Voice Recognition"; |
| const char *AudioHardware::inputPathNameVoiceCommunication = "Voice Communication"; |
| |
| AudioHardware::AudioHardware() : |
| mInit(false), |
| mMicMute(false), |
| mPcm(NULL), |
| mMixer(NULL), |
| mPcmOpenCnt(0), |
| mMixerOpenCnt(0), |
| mInCallAudioMode(false), |
| mVoiceVol(1.0f), |
| mInputSource(AUDIO_SOURCE_DEFAULT), |
| mBluetoothNrec(true), |
| mTTYMode(TTY_MODE_OFF), |
| mSecRilLibHandle(NULL), |
| mRilClient(0), |
| mActivatedCP(false), |
| mDriverOp(DRV_NONE) |
| { |
| loadRILD(); |
| mInit = true; |
| } |
| |
| AudioHardware::~AudioHardware() |
| { |
| for (size_t index = 0; index < mInputs.size(); index++) { |
| closeInputStream(mInputs[index].get()); |
| } |
| mInputs.clear(); |
| closeOutputStream((AudioStreamOut*)mOutput.get()); |
| |
| if (mMixer) { |
| TRACE_DRIVER_IN(DRV_MIXER_CLOSE) |
| mixer_close(mMixer); |
| TRACE_DRIVER_OUT |
| } |
| if (mPcm) { |
| TRACE_DRIVER_IN(DRV_PCM_CLOSE) |
| pcm_close(mPcm); |
| TRACE_DRIVER_OUT |
| } |
| |
| if (mSecRilLibHandle) { |
| if (disconnectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) |
| LOGE("Disconnect_RILD() error"); |
| |
| if (closeClientRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) |
| LOGE("CloseClient_RILD() error"); |
| |
| mRilClient = 0; |
| |
| dlclose(mSecRilLibHandle); |
| mSecRilLibHandle = NULL; |
| } |
| |
| mInit = false; |
| } |
| |
| status_t AudioHardware::initCheck() |
| { |
| return mInit ? NO_ERROR : NO_INIT; |
| } |
| |
| void AudioHardware::loadRILD(void) |
| { |
| mSecRilLibHandle = dlopen("libsecril-client.so", RTLD_NOW); |
| |
| if (mSecRilLibHandle) { |
| LOGV("libsecril-client.so is loaded"); |
| |
| openClientRILD = (HRilClient (*)(void)) |
| dlsym(mSecRilLibHandle, "OpenClient_RILD"); |
| disconnectRILD = (int (*)(HRilClient)) |
| dlsym(mSecRilLibHandle, "Disconnect_RILD"); |
| closeClientRILD = (int (*)(HRilClient)) |
| dlsym(mSecRilLibHandle, "CloseClient_RILD"); |
| isConnectedRILD = (int (*)(HRilClient)) |
| dlsym(mSecRilLibHandle, "isConnected_RILD"); |
| connectRILD = (int (*)(HRilClient)) |
| dlsym(mSecRilLibHandle, "Connect_RILD"); |
| setCallVolume = (int (*)(HRilClient, SoundType, int)) |
| dlsym(mSecRilLibHandle, "SetCallVolume"); |
| setCallAudioPath = (int (*)(HRilClient, AudioPath)) |
| dlsym(mSecRilLibHandle, "SetCallAudioPath"); |
| setCallClockSync = (int (*)(HRilClient, SoundClockCondition)) |
| dlsym(mSecRilLibHandle, "SetCallClockSync"); |
| |
| if (!openClientRILD || !disconnectRILD || !closeClientRILD || |
| !isConnectedRILD || !connectRILD || |
| !setCallVolume || !setCallAudioPath || !setCallClockSync) { |
| LOGE("Can't load all functions from libsecril-client.so"); |
| |
| dlclose(mSecRilLibHandle); |
| mSecRilLibHandle = NULL; |
| } else { |
| mRilClient = openClientRILD(); |
| if (!mRilClient) { |
| LOGE("OpenClient_RILD() error"); |
| |
| dlclose(mSecRilLibHandle); |
| mSecRilLibHandle = NULL; |
| } |
| } |
| } else { |
| LOGE("Can't load libsecril-client.so"); |
| } |
| } |
| |
| status_t AudioHardware::connectRILDIfRequired(void) |
| { |
| if (!mSecRilLibHandle) { |
| LOGE("connectIfRequired() lib is not loaded"); |
| return INVALID_OPERATION; |
| } |
| |
| if (isConnectedRILD(mRilClient)) { |
| return OK; |
| } |
| |
| if (connectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) { |
| LOGE("Connect_RILD() error"); |
| return INVALID_OPERATION; |
| } |
| |
| return OK; |
| } |
| |
| AudioStreamOut* AudioHardware::openOutputStream( |
| uint32_t devices, int *format, uint32_t *channels, |
| uint32_t *sampleRate, status_t *status) |
| { |
| sp <AudioStreamOutALSA> out; |
| status_t rc; |
| |
| { // scope for the lock |
| Mutex::Autolock lock(mLock); |
| |
| // only one output stream allowed |
| if (mOutput != 0) { |
| if (status) { |
| *status = INVALID_OPERATION; |
| } |
| return NULL; |
| } |
| |
| out = new AudioStreamOutALSA(); |
| |
| rc = out->set(this, devices, format, channels, sampleRate); |
| if (rc == NO_ERROR) { |
| mOutput = out; |
| } |
| } |
| |
| if (rc != NO_ERROR) { |
| if (out != 0) { |
| out.clear(); |
| } |
| } |
| if (status) { |
| *status = rc; |
| } |
| |
| return out.get(); |
| } |
| |
| void AudioHardware::closeOutputStream(AudioStreamOut* out) { |
| sp <AudioStreamOutALSA> spOut; |
| { |
| Mutex::Autolock lock(mLock); |
| if (mOutput == 0 || mOutput.get() != out) { |
| LOGW("Attempt to close invalid output stream"); |
| return; |
| } |
| spOut = mOutput; |
| mOutput.clear(); |
| } |
| spOut.clear(); |
| } |
| |
| AudioStreamIn* AudioHardware::openInputStream( |
| uint32_t devices, int *format, uint32_t *channels, |
| uint32_t *sampleRate, status_t *status, |
| AudioSystem::audio_in_acoustics acoustic_flags) |
| { |
| // check for valid input source |
| if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) { |
| if (status) { |
| *status = BAD_VALUE; |
| } |
| return NULL; |
| } |
| |
| status_t rc = NO_ERROR; |
| sp <AudioStreamInALSA> in; |
| |
| { // scope for the lock |
| Mutex::Autolock lock(mLock); |
| |
| in = new AudioStreamInALSA(); |
| rc = in->set(this, devices, format, channels, sampleRate, acoustic_flags); |
| if (rc == NO_ERROR) { |
| mInputs.add(in); |
| } |
| } |
| |
| if (rc != NO_ERROR) { |
| if (in != 0) { |
| in.clear(); |
| } |
| } |
| if (status) { |
| *status = rc; |
| } |
| |
| LOGV("AudioHardware::openInputStream()%p", in.get()); |
| return in.get(); |
| } |
| |
| void AudioHardware::closeInputStream(AudioStreamIn* in) { |
| |
| sp<AudioStreamInALSA> spIn; |
| { |
| Mutex::Autolock lock(mLock); |
| |
| ssize_t index = mInputs.indexOf((AudioStreamInALSA *)in); |
| if (index < 0) { |
| LOGW("Attempt to close invalid input stream"); |
| return; |
| } |
| spIn = mInputs[index]; |
| mInputs.removeAt(index); |
| } |
| LOGV("AudioHardware::closeInputStream()%p", in); |
| spIn.clear(); |
| } |
| |
| |
| status_t AudioHardware::setMode(int mode) |
| { |
| sp<AudioStreamOutALSA> spOut; |
| sp<AudioStreamInALSA> spIn; |
| status_t status; |
| |
| // Mutex acquisition order is always out -> in -> hw |
| AutoMutex lock(mLock); |
| |
| spOut = mOutput; |
| while (spOut != 0) { |
| if (!spOut->checkStandby()) { |
| int cnt = spOut->prepareLock(); |
| mLock.unlock(); |
| spOut->lock(); |
| mLock.lock(); |
| // make sure that another thread did not change output state while the |
| // mutex is released |
| if ((spOut == mOutput) && (cnt == spOut->standbyCnt())) { |
| break; |
| } |
| spOut->unlock(); |
| spOut = mOutput; |
| } else { |
| spOut.clear(); |
| } |
| } |
| // spOut is not 0 here only if the output is active |
| |
| spIn = getActiveInput_l(); |
| while (spIn != 0) { |
| int cnt = spIn->prepareLock(); |
| mLock.unlock(); |
| spIn->lock(); |
| mLock.lock(); |
| // make sure that another thread did not change input state while the |
| // mutex is released |
| if ((spIn == getActiveInput_l()) && (cnt == spIn->standbyCnt())) { |
| break; |
| } |
| spIn->unlock(); |
| spIn = getActiveInput_l(); |
| } |
| // spIn is not 0 here only if the input is active |
| |
| int prevMode = mMode; |
| status = AudioHardwareBase::setMode(mode); |
| LOGV("setMode() : new %d, old %d", mMode, prevMode); |
| if (status == NO_ERROR) { |
| // activate call clock in radio when entering in call or ringtone mode |
| if (prevMode == AudioSystem::MODE_NORMAL) |
| { |
| if ((!mActivatedCP) && (mSecRilLibHandle) && (connectRILDIfRequired() == OK)) { |
| setCallClockSync(mRilClient, SOUND_CLOCK_START); |
| mActivatedCP = true; |
| } |
| } |
| |
| if (mMode == AudioSystem::MODE_IN_CALL && !mInCallAudioMode) { |
| if (spOut != 0) { |
| LOGV("setMode() in call force output standby"); |
| spOut->doStandby_l(); |
| } |
| if (spIn != 0) { |
| LOGV("setMode() in call force input standby"); |
| spIn->doStandby_l(); |
| } |
| |
| LOGV("setMode() openPcmOut_l()"); |
| openPcmOut_l(); |
| openMixer_l(); |
| setInputSource_l(AUDIO_SOURCE_DEFAULT); |
| setVoiceVolume_l(mVoiceVol); |
| mInCallAudioMode = true; |
| } |
| if (mMode == AudioSystem::MODE_NORMAL && mInCallAudioMode) { |
| setInputSource_l(mInputSource); |
| if (mMixer != NULL) { |
| TRACE_DRIVER_IN(DRV_MIXER_GET) |
| struct mixer_ctl *ctl= mixer_get_control(mMixer, "Playback Path", 0); |
| TRACE_DRIVER_OUT |
| if (ctl != NULL) { |
| LOGV("setMode() reset Playback Path to RCV"); |
| TRACE_DRIVER_IN(DRV_MIXER_SEL) |
| mixer_ctl_select(ctl, "RCV"); |
| TRACE_DRIVER_OUT |
| } |
| } |
| LOGV("setMode() closePcmOut_l()"); |
| closeMixer_l(); |
| closePcmOut_l(); |
| |
| if (spOut != 0) { |
| LOGV("setMode() off call force output standby"); |
| spOut->doStandby_l(); |
| } |
| if (spIn != 0) { |
| LOGV("setMode() off call force input standby"); |
| spIn->doStandby_l(); |
| } |
| |
| mInCallAudioMode = false; |
| } |
| |
| if (mMode == AudioSystem::MODE_NORMAL) { |
| if(mActivatedCP) |
| mActivatedCP = false; |
| } |
| } |
| |
| if (spIn != 0) { |
| spIn->unlock(); |
| } |
| if (spOut != 0) { |
| spOut->unlock(); |
| } |
| |
| return status; |
| } |
| |
| status_t AudioHardware::setMicMute(bool state) |
| { |
| LOGV("setMicMute(%d) mMicMute %d", state, mMicMute); |
| sp<AudioStreamInALSA> spIn; |
| { |
| AutoMutex lock(mLock); |
| if (mMicMute != state) { |
| mMicMute = state; |
| // in call mute is handled by RIL |
| if (mMode != AudioSystem::MODE_IN_CALL) { |
| spIn = getActiveInput_l(); |
| } |
| } |
| } |
| |
| if (spIn != 0) { |
| spIn->standby(); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioHardware::getMicMute(bool* state) |
| { |
| *state = mMicMute; |
| return NO_ERROR; |
| } |
| |
| status_t AudioHardware::setParameters(const String8& keyValuePairs) |
| { |
| AudioParameter param = AudioParameter(keyValuePairs); |
| String8 value; |
| String8 key; |
| const char BT_NREC_KEY[] = "bt_headset_nrec"; |
| const char BT_NREC_VALUE_ON[] = "on"; |
| const char TTY_MODE_KEY[] = "tty_mode"; |
| const char TTY_MODE_VALUE_OFF[] = "tty_off"; |
| const char TTY_MODE_VALUE_VCO[] = "tty_vco"; |
| const char TTY_MODE_VALUE_HCO[] = "tty_hco"; |
| const char TTY_MODE_VALUE_FULL[] = "tty_full"; |
| |
| key = String8(BT_NREC_KEY); |
| if (param.get(key, value) == NO_ERROR) { |
| if (value == BT_NREC_VALUE_ON) { |
| mBluetoothNrec = true; |
| } else { |
| mBluetoothNrec = false; |
| LOGD("Turning noise reduction and echo cancellation off for BT " |
| "headset"); |
| } |
| param.remove(String8(BT_NREC_KEY)); |
| } |
| |
| key = String8(TTY_MODE_KEY); |
| if (param.get(key, value) == NO_ERROR) { |
| int ttyMode; |
| if (value == TTY_MODE_VALUE_OFF) { |
| ttyMode = TTY_MODE_OFF; |
| } else if (value == TTY_MODE_VALUE_VCO) { |
| ttyMode = TTY_MODE_VCO; |
| } else if (value == TTY_MODE_VALUE_HCO) { |
| ttyMode = TTY_MODE_HCO; |
| } else if (value == TTY_MODE_VALUE_FULL) { |
| ttyMode = TTY_MODE_FULL; |
| } else { |
| return BAD_VALUE; |
| } |
| |
| if (ttyMode != mTTYMode) { |
| LOGV("new tty mode %d", ttyMode); |
| mTTYMode = ttyMode; |
| if (mOutput != 0 && mMode == AudioSystem::MODE_IN_CALL) { |
| setIncallPath_l(mOutput->device()); |
| } |
| } |
| param.remove(String8(TTY_MODE_KEY)); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| String8 AudioHardware::getParameters(const String8& keys) |
| { |
| AudioParameter request = AudioParameter(keys); |
| AudioParameter reply = AudioParameter(); |
| |
| LOGV("getParameters() %s", keys.string()); |
| |
| return reply.toString(); |
| } |
| |
| size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) |
| { |
| if (format != AudioSystem::PCM_16_BIT) { |
| LOGW("getInputBufferSize bad format: %d", format); |
| return 0; |
| } |
| if (channelCount < 1 || channelCount > 2) { |
| LOGW("getInputBufferSize bad channel count: %d", channelCount); |
| return 0; |
| } |
| if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 && |
| sampleRate != 22050 && sampleRate != 44100) { |
| LOGW("getInputBufferSize bad sample rate: %d", sampleRate); |
| return 0; |
| } |
| |
| return AudioStreamInALSA::getBufferSize(sampleRate, channelCount); |
| } |
| |
| status_t AudioHardware::setVoiceVolume(float volume) |
| { |
| AutoMutex lock(mLock); |
| |
| setVoiceVolume_l(volume); |
| |
| return NO_ERROR; |
| } |
| |
| void AudioHardware::setVoiceVolume_l(float volume) |
| { |
| LOGD("### setVoiceVolume_l"); |
| |
| mVoiceVol = volume; |
| |
| if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) && |
| (connectRILDIfRequired() == OK) ) { |
| |
| uint32_t device = AudioSystem::DEVICE_OUT_EARPIECE; |
| if (mOutput != 0) { |
| device = mOutput->device(); |
| } |
| int int_volume = (int)(volume * 5); |
| SoundType type; |
| |
| LOGD("### route(%d) call volume(%f)", device, volume); |
| switch (device) { |
| case AudioSystem::DEVICE_OUT_EARPIECE: |
| LOGD("### earpiece call volume"); |
| type = SOUND_TYPE_VOICE; |
| break; |
| |
| case AudioSystem::DEVICE_OUT_SPEAKER: |
| LOGD("### speaker call volume"); |
| type = SOUND_TYPE_SPEAKER; |
| break; |
| |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| LOGD("### bluetooth call volume"); |
| type = SOUND_TYPE_BTVOICE; |
| break; |
| |
| case AudioSystem::DEVICE_OUT_WIRED_HEADSET: |
| case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: // Use receive path with 3 pole headset. |
| LOGD("### headset call volume"); |
| type = SOUND_TYPE_HEADSET; |
| break; |
| |
| default: |
| LOGW("### Call volume setting error!!!0x%08x \n", device); |
| type = SOUND_TYPE_VOICE; |
| break; |
| } |
| setCallVolume(mRilClient, type, int_volume); |
| } |
| |
| } |
| |
| status_t AudioHardware::setMasterVolume(float volume) |
| { |
| LOGV("Set master volume to %f.\n", volume); |
| // We return an error code here to let the audioflinger do in-software |
| // volume on top of the maximum volume that we set through the SND API. |
| // return error - software mixer will handle it |
| return -1; |
| } |
| |
| static const int kDumpLockRetries = 50; |
| static const int kDumpLockSleep = 20000; |
| |
| static bool tryLock(Mutex& mutex) |
| { |
| bool locked = false; |
| for (int i = 0; i < kDumpLockRetries; ++i) { |
| if (mutex.tryLock() == NO_ERROR) { |
| locked = true; |
| break; |
| } |
| usleep(kDumpLockSleep); |
| } |
| return locked; |
| } |
| |
| status_t AudioHardware::dump(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| bool locked = tryLock(mLock); |
| if (!locked) { |
| snprintf(buffer, SIZE, "\n\tAudioHardware maybe deadlocked\n"); |
| } else { |
| mLock.unlock(); |
| } |
| |
| snprintf(buffer, SIZE, "\tInit %s\n", (mInit) ? "OK" : "Failed"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tMic Mute %s\n", (mMicMute) ? "ON" : "OFF"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmPcm: %p\n", mPcm); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmPcmOpenCnt: %d\n", mPcmOpenCnt); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmMixer: %p\n", mMixer); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmMixerOpenCnt: %d\n", mMixerOpenCnt); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tIn Call Audio Mode %s\n", |
| (mInCallAudioMode) ? "ON" : "OFF"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tInput source %d\n", mInputSource); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmSecRilLibHandle: %p\n", mSecRilLibHandle); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmRilClient: %p\n", mRilClient); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tCP %s\n", |
| (mActivatedCP) ? "Activated" : "Deactivated"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\tmDriverOp: %d\n", mDriverOp); |
| result.append(buffer); |
| |
| snprintf(buffer, SIZE, "\n\tmOutput %p dump:\n", mOutput.get()); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| if (mOutput != 0) { |
| mOutput->dump(fd, args); |
| } |
| |
| snprintf(buffer, SIZE, "\n\t%d inputs opened:\n", mInputs.size()); |
| write(fd, buffer, strlen(buffer)); |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| snprintf(buffer, SIZE, "\t- input %d dump:\n", i); |
| write(fd, buffer, strlen(buffer)); |
| mInputs[i]->dump(fd, args); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioHardware::setIncallPath_l(uint32_t device) |
| { |
| LOGV("setIncallPath_l: device %x", device); |
| |
| // Setup sound path for CP clocking |
| if ((mSecRilLibHandle) && |
| (connectRILDIfRequired() == OK)) { |
| |
| if (mMode == AudioSystem::MODE_IN_CALL) { |
| LOGD("### incall mode route (%d)", device); |
| AudioPath path; |
| switch(device){ |
| case AudioSystem::DEVICE_OUT_EARPIECE: |
| LOGD("### incall mode earpiece route"); |
| path = SOUND_AUDIO_PATH_HANDSET; |
| break; |
| |
| case AudioSystem::DEVICE_OUT_SPEAKER: |
| LOGD("### incall mode speaker route"); |
| path = SOUND_AUDIO_PATH_SPEAKER; |
| break; |
| |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| LOGD("### incall mode bluetooth route %s NR", mBluetoothNrec ? "" : "NO"); |
| if (mBluetoothNrec) { |
| path = SOUND_AUDIO_PATH_BLUETOOTH; |
| } else { |
| path = SOUND_AUDIO_PATH_BLUETOOTH_NO_NR; |
| } |
| break; |
| |
| case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE : |
| LOGD("### incall mode headphone route"); |
| path = SOUND_AUDIO_PATH_HEADPHONE; |
| break; |
| case AudioSystem::DEVICE_OUT_WIRED_HEADSET : |
| LOGD("### incall mode headset route"); |
| path = SOUND_AUDIO_PATH_HEADSET; |
| break; |
| default: |
| LOGW("### incall mode Error!! route = [%d]", device); |
| path = SOUND_AUDIO_PATH_HANDSET; |
| break; |
| } |
| |
| setCallAudioPath(mRilClient, path); |
| |
| if (mMixer != NULL) { |
| TRACE_DRIVER_IN(DRV_MIXER_GET) |
| struct mixer_ctl *ctl= mixer_get_control(mMixer, "Voice Call Path", 0); |
| TRACE_DRIVER_OUT |
| LOGE_IF(ctl == NULL, "setIncallPath_l() could not get mixer ctl"); |
| if (ctl != NULL) { |
| LOGV("setIncallPath_l() Voice Call Path, (%x)", device); |
| TRACE_DRIVER_IN(DRV_MIXER_SEL) |
| mixer_ctl_select(ctl, getVoiceRouteFromDevice(device)); |
| TRACE_DRIVER_OUT |
| } |
| } |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| struct pcm *AudioHardware::openPcmOut_l() |
| { |
| LOGD("openPcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); |
| if (mPcmOpenCnt++ == 0) { |
| if (mPcm != NULL) { |
| LOGE("openPcmOut_l() mPcmOpenCnt == 0 and mPcm == %p\n", mPcm); |
| mPcmOpenCnt--; |
| return NULL; |
| } |
| unsigned flags = PCM_OUT; |
| |
| flags |= (AUDIO_HW_OUT_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT; |
| flags |= (AUDIO_HW_OUT_PERIOD_CNT - PCM_PERIOD_CNT_MIN) << PCM_PERIOD_CNT_SHIFT; |
| |
| TRACE_DRIVER_IN(DRV_PCM_OPEN) |
| mPcm = pcm_open(flags); |
| TRACE_DRIVER_OUT |
| if (!pcm_ready(mPcm)) { |
| LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_error(mPcm)); |
| TRACE_DRIVER_IN(DRV_PCM_CLOSE) |
| pcm_close(mPcm); |
| TRACE_DRIVER_OUT |
| mPcmOpenCnt--; |
| mPcm = NULL; |
| } |
| } |
| return mPcm; |
| } |
| |
| void AudioHardware::closePcmOut_l() |
| { |
| LOGD("closePcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); |
| if (mPcmOpenCnt == 0) { |
| LOGE("closePcmOut_l() mPcmOpenCnt == 0"); |
| return; |
| } |
| |
| if (--mPcmOpenCnt == 0) { |
| TRACE_DRIVER_IN(DRV_PCM_CLOSE) |
| pcm_close(mPcm); |
| TRACE_DRIVER_OUT |
| mPcm = NULL; |
| } |
| } |
| |
| struct mixer *AudioHardware::openMixer_l() |
| { |
| LOGV("openMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); |
| if (mMixerOpenCnt++ == 0) { |
| if (mMixer != NULL) { |
| LOGE("openMixer_l() mMixerOpenCnt == 0 and mMixer == %p\n", mMixer); |
| mMixerOpenCnt--; |
| return NULL; |
| } |
| TRACE_DRIVER_IN(DRV_MIXER_OPEN) |
| mMixer = mixer_open(); |
| TRACE_DRIVER_OUT |
| if (mMixer == NULL) { |
| LOGE("openMixer_l() cannot open mixer"); |
| mMixerOpenCnt--; |
| return NULL; |
| } |
| } |
| return mMixer; |
| } |
| |
| void AudioHardware::closeMixer_l() |
| { |
| LOGV("closeMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); |
| if (mMixerOpenCnt == 0) { |
| LOGE("closeMixer_l() mMixerOpenCnt == 0"); |
| return; |
| } |
| |
| if (--mMixerOpenCnt == 0) { |
| TRACE_DRIVER_IN(DRV_MIXER_CLOSE) |
| mixer_close(mMixer); |
| TRACE_DRIVER_OUT |
| mMixer = NULL; |
| } |
| } |
| |
| const char *AudioHardware::getOutputRouteFromDevice(uint32_t device) |
| { |
| switch (device) { |
| case AudioSystem::DEVICE_OUT_EARPIECE: |
| return "RCV"; |
| case AudioSystem::DEVICE_OUT_SPEAKER: |
| if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK"; |
| else return "SPK"; |
| case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: |
| if (mMode == AudioSystem::MODE_RINGTONE) return "RING_NO_MIC"; |
| else return "HP_NO_MIC"; |
| case AudioSystem::DEVICE_OUT_WIRED_HEADSET: |
| if (mMode == AudioSystem::MODE_RINGTONE) return "RING_HP"; |
| else return "HP"; |
| case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADPHONE): |
| case (AudioSystem::DEVICE_OUT_SPEAKER|AudioSystem::DEVICE_OUT_WIRED_HEADSET): |
| if (mMode == AudioSystem::MODE_RINGTONE) return "RING_SPK_HP"; |
| else return "SPK_HP"; |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| return "BT"; |
| default: |
| return "OFF"; |
| } |
| } |
| |
| const char *AudioHardware::getVoiceRouteFromDevice(uint32_t device) |
| { |
| switch (device) { |
| case AudioSystem::DEVICE_OUT_EARPIECE: |
| return "RCV"; |
| case AudioSystem::DEVICE_OUT_SPEAKER: |
| return "SPK"; |
| case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: |
| case AudioSystem::DEVICE_OUT_WIRED_HEADSET: |
| switch (mTTYMode) { |
| case TTY_MODE_VCO: |
| return "TTY_VCO"; |
| case TTY_MODE_HCO: |
| return "TTY_HCO"; |
| case TTY_MODE_FULL: |
| return "TTY_FULL"; |
| case TTY_MODE_OFF: |
| default: |
| if (device == AudioSystem::DEVICE_OUT_WIRED_HEADPHONE) { |
| return "HP_NO_MIC"; |
| } else { |
| return "HP"; |
| } |
| } |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: |
| case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: |
| return "BT"; |
| default: |
| return "OFF"; |
| } |
| } |
| |
| const char *AudioHardware::getInputRouteFromDevice(uint32_t device) |
| { |
| if (mMicMute) { |
| return "MIC OFF"; |
| } |
| |
| switch (device) { |
| case AudioSystem::DEVICE_IN_BUILTIN_MIC: |
| return "Main Mic"; |
| case AudioSystem::DEVICE_IN_WIRED_HEADSET: |
| return "Hands Free Mic"; |
| case AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET: |
| return "BT Sco Mic"; |
| default: |
| return "MIC OFF"; |
| } |
| } |
| |
| uint32_t AudioHardware::getInputSampleRate(uint32_t sampleRate) |
| { |
| uint32_t i; |
| uint32_t prevDelta; |
| uint32_t delta; |
| |
| for (i = 0, prevDelta = 0xFFFFFFFF; i < sizeof(inputSamplingRates)/sizeof(uint32_t); i++, prevDelta = delta) { |
| delta = abs(sampleRate - inputSamplingRates[i]); |
| if (delta > prevDelta) break; |
| } |
| // i is always > 0 here |
| return inputSamplingRates[i-1]; |
| } |
| |
| // getActiveInput_l() must be called with mLock held |
| sp <AudioHardware::AudioStreamInALSA> AudioHardware::getActiveInput_l() |
| { |
| sp< AudioHardware::AudioStreamInALSA> spIn; |
| |
| for (size_t i = 0; i < mInputs.size(); i++) { |
| // return first input found not being in standby mode |
| // as only one input can be in this state |
| if (!mInputs[i]->checkStandby()) { |
| spIn = mInputs[i]; |
| break; |
| } |
| } |
| |
| return spIn; |
| } |
| |
| status_t AudioHardware::setInputSource_l(audio_source source) |
| { |
| LOGV("setInputSource_l(%d)", source); |
| if (source != mInputSource) { |
| if ((source == AUDIO_SOURCE_DEFAULT) || (mMode != AudioSystem::MODE_IN_CALL)) { |
| if (mMixer) { |
| TRACE_DRIVER_IN(DRV_MIXER_GET) |
| struct mixer_ctl *ctl= mixer_get_control(mMixer, "Input Source", 0); |
| TRACE_DRIVER_OUT |
| if (ctl == NULL) { |
| return NO_INIT; |
| } |
| const char* sourceName; |
| switch (source) { |
| case AUDIO_SOURCE_DEFAULT: // intended fall-through |
| case AUDIO_SOURCE_MIC: |
| sourceName = inputPathNameDefault; |
| break; |
| case AUDIO_SOURCE_VOICE_COMMUNICATION: |
| sourceName = inputPathNameVoiceCommunication; |
| break; |
| case AUDIO_SOURCE_CAMCORDER: |
| sourceName = inputPathNameCamcorder; |
| break; |
| case AUDIO_SOURCE_VOICE_RECOGNITION: |
| sourceName = inputPathNameVoiceRecognition; |
| break; |
| case AUDIO_SOURCE_VOICE_UPLINK: // intended fall-through |
| case AUDIO_SOURCE_VOICE_DOWNLINK: // intended fall-through |
| case AUDIO_SOURCE_VOICE_CALL: // intended fall-through |
| default: |
| return NO_INIT; |
| } |
| LOGV("mixer_ctl_select, Input Source, (%s)", sourceName); |
| TRACE_DRIVER_IN(DRV_MIXER_SEL) |
| mixer_ctl_select(ctl, sourceName); |
| TRACE_DRIVER_OUT |
| } |
| } |
| mInputSource = source; |
| } |
| |
| return NO_ERROR; |
| } |
| |
| |
| //------------------------------------------------------------------------------ |
| // AudioStreamOutALSA |
| //------------------------------------------------------------------------------ |
| |
| AudioHardware::AudioStreamOutALSA::AudioStreamOutALSA() : |
| mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0), |
| mStandby(true), mDevices(0), mChannels(AUDIO_HW_OUT_CHANNELS), |
| mSampleRate(AUDIO_HW_OUT_SAMPLERATE), mBufferSize(AUDIO_HW_OUT_PERIOD_BYTES), |
| mDriverOp(DRV_NONE), mStandbyCnt(0), mSleepReq(false) |
| { |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::set( |
| AudioHardware* hw, uint32_t devices, int *pFormat, |
| uint32_t *pChannels, uint32_t *pRate) |
| { |
| int lFormat = pFormat ? *pFormat : 0; |
| uint32_t lChannels = pChannels ? *pChannels : 0; |
| uint32_t lRate = pRate ? *pRate : 0; |
| |
| mHardware = hw; |
| mDevices = devices; |
| |
| // fix up defaults |
| if (lFormat == 0) lFormat = format(); |
| if (lChannels == 0) lChannels = channels(); |
| if (lRate == 0) lRate = sampleRate(); |
| |
| // check values |
| if ((lFormat != format()) || |
| (lChannels != channels()) || |
| (lRate != sampleRate())) { |
| if (pFormat) *pFormat = format(); |
| if (pChannels) *pChannels = channels(); |
| if (pRate) *pRate = sampleRate(); |
| return BAD_VALUE; |
| } |
| |
| if (pFormat) *pFormat = lFormat; |
| if (pChannels) *pChannels = lChannels; |
| if (pRate) *pRate = lRate; |
| |
| mChannels = lChannels; |
| mSampleRate = lRate; |
| mBufferSize = AUDIO_HW_OUT_PERIOD_BYTES; |
| |
| return NO_ERROR; |
| } |
| |
| AudioHardware::AudioStreamOutALSA::~AudioStreamOutALSA() |
| { |
| standby(); |
| } |
| |
| ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes) |
| { |
| // LOGV("AudioStreamOutALSA::write(%p, %u)", buffer, bytes); |
| status_t status = NO_INIT; |
| const uint8_t* p = static_cast<const uint8_t*>(buffer); |
| int ret; |
| |
| if (mHardware == NULL) return NO_INIT; |
| |
| if (mSleepReq) { |
| // 10ms are always shorter than the time to reconfigure the audio path |
| // which is the only condition when mSleepReq would be true. |
| usleep(10000); |
| } |
| |
| { // scope for the lock |
| |
| AutoMutex lock(mLock); |
| |
| if (mStandby) { |
| AutoMutex hwLock(mHardware->lock()); |
| |
| LOGD("AudioHardware pcm playback is exiting standby."); |
| acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock"); |
| |
| sp<AudioStreamInALSA> spIn = mHardware->getActiveInput_l(); |
| while (spIn != 0) { |
| int cnt = spIn->prepareLock(); |
| mHardware->lock().unlock(); |
| // Mutex acquisition order is always out -> in -> hw |
| spIn->lock(); |
| mHardware->lock().lock(); |
| // make sure that another thread did not change input state |
| // while the mutex is released |
| if ((spIn == mHardware->getActiveInput_l()) && |
| (cnt == spIn->standbyCnt())) { |
| LOGV("AudioStreamOutALSA::write() force input standby"); |
| spIn->close_l(); |
| break; |
| } |
| spIn->unlock(); |
| spIn = mHardware->getActiveInput_l(); |
| } |
| // spIn is not 0 here only if the input was active and has been |
| // closed above |
| |
| // open output before input |
| open_l(); |
| |
| if (spIn != 0) { |
| if (spIn->open_l() != NO_ERROR) { |
| spIn->doStandby_l(); |
| } |
| spIn->unlock(); |
| } |
| if (mPcm == NULL) { |
| release_wake_lock("AudioOutLock"); |
| goto Error; |
| } |
| mStandby = false; |
| } |
| |
| TRACE_DRIVER_IN(DRV_PCM_WRITE) |
| ret = pcm_write(mPcm,(void*) p, bytes); |
| TRACE_DRIVER_OUT |
| |
| if (ret == 0) { |
| return bytes; |
| } |
| LOGW("write error: %d", errno); |
| status = -errno; |
| } |
| Error: |
| |
| standby(); |
| |
| // Simulate audio output timing in case of error |
| usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate()); |
| |
| return status; |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::standby() |
| { |
| if (mHardware == NULL) return NO_INIT; |
| |
| AutoMutex lock(mLock); |
| |
| { // scope for the AudioHardware lock |
| AutoMutex hwLock(mHardware->lock()); |
| |
| doStandby_l(); |
| } |
| |
| return NO_ERROR; |
| } |
| |
| void AudioHardware::AudioStreamOutALSA::doStandby_l() |
| { |
| mStandbyCnt++; |
| |
| if (!mStandby) { |
| LOGD("AudioHardware pcm playback is going to standby."); |
| release_wake_lock("AudioOutLock"); |
| mStandby = true; |
| } |
| |
| close_l(); |
| } |
| |
| void AudioHardware::AudioStreamOutALSA::close_l() |
| { |
| if (mMixer) { |
| mHardware->closeMixer_l(); |
| mMixer = NULL; |
| mRouteCtl = NULL; |
| } |
| if (mPcm) { |
| mHardware->closePcmOut_l(); |
| mPcm = NULL; |
| } |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::open_l() |
| { |
| LOGV("open pcm_out driver"); |
| mPcm = mHardware->openPcmOut_l(); |
| if (mPcm == NULL) { |
| return NO_INIT; |
| } |
| |
| mMixer = mHardware->openMixer_l(); |
| if (mMixer) { |
| LOGV("open playback normal"); |
| TRACE_DRIVER_IN(DRV_MIXER_GET) |
| mRouteCtl = mixer_get_control(mMixer, "Playback Path", 0); |
| TRACE_DRIVER_OUT |
| } |
| if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { |
| const char *route = mHardware->getOutputRouteFromDevice(mDevices); |
| LOGV("write() wakeup setting route %s", route); |
| if (mRouteCtl) { |
| TRACE_DRIVER_IN(DRV_MIXER_SEL) |
| mixer_ctl_select(mRouteCtl, route); |
| TRACE_DRIVER_OUT |
| } |
| } |
| return NO_ERROR; |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::dump(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| bool locked = tryLock(mLock); |
| if (!locked) { |
| snprintf(buffer, SIZE, "\n\t\tAudioStreamOutALSA maybe deadlocked\n"); |
| } else { |
| mLock.unlock(); |
| } |
| |
| snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmRouteCtl: %p\n", mRouteCtl); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp); |
| result.append(buffer); |
| |
| ::write(fd, result.string(), result.size()); |
| |
| return NO_ERROR; |
| } |
| |
| bool AudioHardware::AudioStreamOutALSA::checkStandby() |
| { |
| return mStandby; |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValuePairs) |
| { |
| AudioParameter param = AudioParameter(keyValuePairs); |
| status_t status = NO_ERROR; |
| int device; |
| LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string()); |
| |
| if (mHardware == NULL) return NO_INIT; |
| |
| { |
| AutoMutex lock(mLock); |
| |
| if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR) |
| { |
| if (device != 0) { |
| AutoMutex hwLock(mHardware->lock()); |
| |
| if (mDevices != (uint32_t)device) { |
| mDevices = (uint32_t)device; |
| if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { |
| doStandby_l(); |
| } |
| } |
| if (mHardware->mode() == AudioSystem::MODE_IN_CALL) { |
| mHardware->setIncallPath_l(device); |
| } |
| } |
| param.remove(String8(AudioParameter::keyRouting)); |
| } |
| } |
| |
| if (param.size()) { |
| status = BAD_VALUE; |
| } |
| |
| |
| return status; |
| |
| } |
| |
| String8 AudioHardware::AudioStreamOutALSA::getParameters(const String8& keys) |
| { |
| AudioParameter param = AudioParameter(keys); |
| String8 value; |
| String8 key = String8(AudioParameter::keyRouting); |
| |
| if (param.get(key, value) == NO_ERROR) { |
| param.addInt(key, (int)mDevices); |
| } |
| |
| LOGV("AudioStreamOutALSA::getParameters() %s", param.toString().string()); |
| return param.toString(); |
| } |
| |
| status_t AudioHardware::AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames) |
| { |
| //TODO |
| return INVALID_OPERATION; |
| } |
| |
| int AudioHardware::AudioStreamOutALSA::prepareLock() |
| { |
| // request sleep next time write() is called so that caller can acquire |
| // mLock |
| mSleepReq = true; |
| return mStandbyCnt; |
| } |
| |
| void AudioHardware::AudioStreamOutALSA::lock() |
| { |
| mLock.lock(); |
| mSleepReq = false; |
| } |
| |
| void AudioHardware::AudioStreamOutALSA::unlock() { |
| mLock.unlock(); |
| } |
| |
| //------------------------------------------------------------------------------ |
| // AudioStreamInALSA |
| //------------------------------------------------------------------------------ |
| |
| AudioHardware::AudioStreamInALSA::AudioStreamInALSA() : |
| mHardware(0), mPcm(0), mMixer(0), mRouteCtl(0), |
| mStandby(true), mDevices(0), mChannels(AUDIO_HW_IN_CHANNELS), mChannelCount(1), |
| mSampleRate(AUDIO_HW_IN_SAMPLERATE), mBufferSize(AUDIO_HW_IN_PERIOD_BYTES), |
| mDownSampler(NULL), mReadStatus(NO_ERROR), mDriverOp(DRV_NONE), |
| mStandbyCnt(0), mSleepReq(false) |
| { |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::set( |
| AudioHardware* hw, uint32_t devices, int *pFormat, |
| uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics) |
| { |
| if (pFormat == 0 || *pFormat != AUDIO_HW_IN_FORMAT) { |
| *pFormat = AUDIO_HW_IN_FORMAT; |
| return BAD_VALUE; |
| } |
| if (pRate == 0) { |
| return BAD_VALUE; |
| } |
| uint32_t rate = AudioHardware::getInputSampleRate(*pRate); |
| if (rate != *pRate) { |
| *pRate = rate; |
| return BAD_VALUE; |
| } |
| |
| if (pChannels == 0 || (*pChannels != AudioSystem::CHANNEL_IN_MONO && |
| *pChannels != AudioSystem::CHANNEL_IN_STEREO)) { |
| *pChannels = AUDIO_HW_IN_CHANNELS; |
| return BAD_VALUE; |
| } |
| |
| mHardware = hw; |
| |
| LOGV("AudioStreamInALSA::set(%d, %d, %u)", *pFormat, *pChannels, *pRate); |
| |
| mBufferSize = getBufferSize(*pRate, AudioSystem::popCount(*pChannels)); |
| mDevices = devices; |
| mChannels = *pChannels; |
| mChannelCount = AudioSystem::popCount(mChannels); |
| mSampleRate = rate; |
| if (mSampleRate != AUDIO_HW_OUT_SAMPLERATE) { |
| mDownSampler = new AudioHardware::DownSampler(mSampleRate, |
| mChannelCount, |
| AUDIO_HW_IN_PERIOD_SZ, |
| this); |
| status_t status = mDownSampler->initCheck(); |
| if (status != NO_ERROR) { |
| delete mDownSampler; |
| LOGW("AudioStreamInALSA::set() downsampler init failed: %d", status); |
| return status; |
| } |
| |
| mPcmIn = new int16_t[AUDIO_HW_IN_PERIOD_SZ * mChannelCount]; |
| } |
| return NO_ERROR; |
| } |
| |
| AudioHardware::AudioStreamInALSA::~AudioStreamInALSA() |
| { |
| standby(); |
| if (mDownSampler != NULL) { |
| delete mDownSampler; |
| if (mPcmIn != NULL) { |
| delete[] mPcmIn; |
| } |
| } |
| } |
| |
| ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes) |
| { |
| // LOGV("AudioStreamInALSA::read(%p, %u)", buffer, bytes); |
| status_t status = NO_INIT; |
| int ret; |
| |
| if (mHardware == NULL) return NO_INIT; |
| |
| if (mSleepReq) { |
| // 10ms are always shorter than the time to reconfigure the audio path |
| // which is the only condition when mSleepReq would be true. |
| usleep(10000); |
| } |
| |
| { // scope for the lock |
| AutoMutex lock(mLock); |
| |
| if (mStandby) { |
| AutoMutex hwLock(mHardware->lock()); |
| |
| LOGD("AudioHardware pcm capture is exiting standby."); |
| acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock"); |
| |
| sp<AudioStreamOutALSA> spOut = mHardware->output(); |
| while (spOut != 0) { |
| if (!spOut->checkStandby()) { |
| int cnt = spOut->prepareLock(); |
| mHardware->lock().unlock(); |
| mLock.unlock(); |
| // Mutex acquisition order is always out -> in -> hw |
| spOut->lock(); |
| mLock.lock(); |
| mHardware->lock().lock(); |
| // make sure that another thread did not change output state |
| // while the mutex is released |
| if ((spOut == mHardware->output()) && (cnt == spOut->standbyCnt())) { |
| LOGV("AudioStreamInALSA::read() force output standby"); |
| spOut->close_l(); |
| break; |
| } |
| spOut->unlock(); |
| spOut = mHardware->output(); |
| } else { |
| spOut.clear(); |
| } |
| } |
| // spOut is not 0 here only if the output was active and has been |
| // closed above |
| |
| // open output before input |
| if (spOut != 0) { |
| if (spOut->open_l() != NO_ERROR) { |
| spOut->doStandby_l(); |
| } |
| spOut->unlock(); |
| } |
| |
| open_l(); |
| |
| if (mPcm == NULL) { |
| release_wake_lock("AudioInLock"); |
| goto Error; |
| } |
| mStandby = false; |
| } |
| |
| |
| if (mDownSampler != NULL) { |
| size_t frames = bytes / frameSize(); |
| size_t framesIn = 0; |
| mReadStatus = 0; |
| do { |
| size_t outframes = frames - framesIn; |
| mDownSampler->resample( |
| (int16_t *)buffer + (framesIn * mChannelCount), |
| &outframes); |
| framesIn += outframes; |
| } while ((framesIn < frames) && mReadStatus == 0); |
| ret = mReadStatus; |
| bytes = framesIn * frameSize(); |
| } else { |
| TRACE_DRIVER_IN(DRV_PCM_READ) |
| ret = pcm_read(mPcm, buffer, bytes); |
| TRACE_DRIVER_OUT |
| } |
| |
| if (ret == 0) { |
| return bytes; |
| } |
| |
| LOGW("read error: %d", ret); |
| status = ret; |
| } |
| |
| Error: |
| |
| standby(); |
| |
| // Simulate audio output timing in case of error |
| usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate()); |
| |
| return status; |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::standby() |
| { |
| if (mHardware == NULL) return NO_INIT; |
| |
| AutoMutex lock(mLock); |
| |
| { // scope for AudioHardware lock |
| AutoMutex hwLock(mHardware->lock()); |
| |
| doStandby_l(); |
| } |
| return NO_ERROR; |
| } |
| |
| void AudioHardware::AudioStreamInALSA::doStandby_l() |
| { |
| mStandbyCnt++; |
| |
| if (!mStandby) { |
| LOGD("AudioHardware pcm capture is going to standby."); |
| release_wake_lock("AudioInLock"); |
| mStandby = true; |
| } |
| close_l(); |
| } |
| |
| void AudioHardware::AudioStreamInALSA::close_l() |
| { |
| if (mMixer) { |
| mHardware->closeMixer_l(); |
| mMixer = NULL; |
| mRouteCtl = NULL; |
| } |
| |
| if (mPcm) { |
| TRACE_DRIVER_IN(DRV_PCM_CLOSE) |
| pcm_close(mPcm); |
| TRACE_DRIVER_OUT |
| mPcm = NULL; |
| } |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::open_l() |
| { |
| unsigned flags = PCM_IN; |
| if (mChannels == AudioSystem::CHANNEL_IN_MONO) { |
| flags |= PCM_MONO; |
| } |
| flags |= (AUDIO_HW_IN_PERIOD_MULT - 1) << PCM_PERIOD_SZ_SHIFT; |
| flags |= (AUDIO_HW_IN_PERIOD_CNT - PCM_PERIOD_CNT_MIN) |
| << PCM_PERIOD_CNT_SHIFT; |
| |
| LOGV("open pcm_in driver"); |
| TRACE_DRIVER_IN(DRV_PCM_OPEN) |
| mPcm = pcm_open(flags); |
| TRACE_DRIVER_OUT |
| if (!pcm_ready(mPcm)) { |
| LOGE("cannot open pcm_in driver: %s\n", pcm_error(mPcm)); |
| TRACE_DRIVER_IN(DRV_PCM_CLOSE) |
| pcm_close(mPcm); |
| TRACE_DRIVER_OUT |
| mPcm = NULL; |
| return NO_INIT; |
| } |
| |
| if (mDownSampler != NULL) { |
| mInPcmInBuf = 0; |
| mDownSampler->reset(); |
| } |
| |
| mMixer = mHardware->openMixer_l(); |
| if (mMixer) { |
| TRACE_DRIVER_IN(DRV_MIXER_GET) |
| mRouteCtl = mixer_get_control(mMixer, "Capture MIC Path", 0); |
| TRACE_DRIVER_OUT |
| } |
| |
| if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { |
| const char *route = mHardware->getInputRouteFromDevice(mDevices); |
| LOGV("read() wakeup setting route %s", route); |
| if (mRouteCtl) { |
| TRACE_DRIVER_IN(DRV_MIXER_SEL) |
| mixer_ctl_select(mRouteCtl, route); |
| TRACE_DRIVER_OUT |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::dump(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| bool locked = tryLock(mLock); |
| if (!locked) { |
| snprintf(buffer, SIZE, "\n\t\tAudioStreamInALSA maybe deadlocked\n"); |
| } else { |
| mLock.unlock(); |
| } |
| |
| snprintf(buffer, SIZE, "\t\tmHardware: %p\n", mHardware); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmPcm: %p\n", mPcm); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmMixer: %p\n", mMixer); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tStandby %s\n", (mStandby) ? "ON" : "OFF"); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmDevices: 0x%08x\n", mDevices); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmChannels: 0x%08x\n", mChannels); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmSampleRate: %d\n", mSampleRate); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmBufferSize: %d\n", mBufferSize); |
| result.append(buffer); |
| snprintf(buffer, SIZE, "\t\tmDriverOp: %d\n", mDriverOp); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| |
| return NO_ERROR; |
| } |
| |
| bool AudioHardware::AudioStreamInALSA::checkStandby() |
| { |
| return mStandby; |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::setParameters(const String8& keyValuePairs) |
| { |
| AudioParameter param = AudioParameter(keyValuePairs); |
| status_t status = NO_ERROR; |
| int value; |
| |
| LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); |
| |
| if (mHardware == NULL) return NO_INIT; |
| |
| { |
| AutoMutex lock(mLock); |
| |
| if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR) { |
| AutoMutex hwLock(mHardware->lock()); |
| |
| mHardware->openMixer_l(); |
| mHardware->setInputSource_l((audio_source)value); |
| mHardware->closeMixer_l(); |
| |
| param.remove(String8(AudioParameter::keyInputSource)); |
| } |
| |
| if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) |
| { |
| if (value != 0) { |
| AutoMutex hwLock(mHardware->lock()); |
| |
| if (mDevices != (uint32_t)value) { |
| mDevices = (uint32_t)value; |
| if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { |
| doStandby_l(); |
| } |
| } |
| } |
| param.remove(String8(AudioParameter::keyRouting)); |
| } |
| } |
| |
| |
| if (param.size()) { |
| status = BAD_VALUE; |
| } |
| |
| return status; |
| |
| } |
| |
| String8 AudioHardware::AudioStreamInALSA::getParameters(const String8& keys) |
| { |
| AudioParameter param = AudioParameter(keys); |
| String8 value; |
| String8 key = String8(AudioParameter::keyRouting); |
| |
| if (param.get(key, value) == NO_ERROR) { |
| param.addInt(key, (int)mDevices); |
| } |
| |
| LOGV("AudioStreamInALSA::getParameters() %s", param.toString().string()); |
| return param.toString(); |
| } |
| |
| status_t AudioHardware::AudioStreamInALSA::getNextBuffer(AudioHardware::BufferProvider::Buffer* buffer) |
| { |
| if (mPcm == NULL) { |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| mReadStatus = NO_INIT; |
| return NO_INIT; |
| } |
| |
| if (mInPcmInBuf == 0) { |
| TRACE_DRIVER_IN(DRV_PCM_READ) |
| mReadStatus = pcm_read(mPcm,(void*) mPcmIn, AUDIO_HW_IN_PERIOD_SZ * frameSize()); |
| TRACE_DRIVER_OUT |
| if (mReadStatus != 0) { |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| return mReadStatus; |
| } |
| mInPcmInBuf = AUDIO_HW_IN_PERIOD_SZ; |
| } |
| |
| buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount; |
| buffer->i16 = mPcmIn + (AUDIO_HW_IN_PERIOD_SZ - mInPcmInBuf) * mChannelCount; |
| |
| return mReadStatus; |
| } |
| |
| void AudioHardware::AudioStreamInALSA::releaseBuffer(Buffer* buffer) |
| { |
| mInPcmInBuf -= buffer->frameCount; |
| } |
| |
| size_t AudioHardware::AudioStreamInALSA::getBufferSize(uint32_t sampleRate, int channelCount) |
| { |
| size_t ratio; |
| |
| switch (sampleRate) { |
| case 8000: |
| case 11025: |
| ratio = 4; |
| break; |
| case 16000: |
| case 22050: |
| ratio = 2; |
| break; |
| case 44100: |
| default: |
| ratio = 1; |
| break; |
| } |
| |
| return (AUDIO_HW_IN_PERIOD_SZ*channelCount*sizeof(int16_t)) / ratio ; |
| } |
| |
| int AudioHardware::AudioStreamInALSA::prepareLock() |
| { |
| // request sleep next time read() is called so that caller can acquire |
| // mLock |
| mSleepReq = true; |
| return mStandbyCnt; |
| } |
| |
| void AudioHardware::AudioStreamInALSA::lock() |
| { |
| mLock.lock(); |
| mSleepReq = false; |
| } |
| |
| void AudioHardware::AudioStreamInALSA::unlock() { |
| mLock.unlock(); |
| } |
| |
| //------------------------------------------------------------------------------ |
| // DownSampler |
| //------------------------------------------------------------------------------ |
| |
| /* |
| * 2.30 fixed point FIR filter coefficients for conversion 44100 -> 22050. |
| * (Works equivalently for 22010 -> 11025 or any other halving, of course.) |
| * |
| * Transition band from about 18 kHz, passband ripple < 0.1 dB, |
| * stopband ripple at about -55 dB, linear phase. |
| * |
| * Design and display in MATLAB or Octave using: |
| * |
| * filter = fir1(19, 0.5); filter = round(filter * 2**30); freqz(filter * 2**-30); |
| */ |
| static const int32_t filter_22khz_coeff[] = { |
| 2089257, 2898328, -5820678, -10484531, |
| 19038724, 30542725, -50469415, -81505260, |
| 152544464, 478517512, 478517512, 152544464, |
| -81505260, -50469415, 30542725, 19038724, |
| -10484531, -5820678, 2898328, 2089257, |
| }; |
| #define NUM_COEFF_22KHZ (sizeof(filter_22khz_coeff) / sizeof(filter_22khz_coeff[0])) |
| #define OVERLAP_22KHZ (NUM_COEFF_22KHZ - 2) |
| |
| /* |
| * Convolution of signals A and reverse(B). (In our case, the filter response |
| * is symmetric, so the reversing doesn't matter.) |
| * A is taken to be in 0.16 fixed-point, and B is taken to be in 2.30 fixed-point. |
| * The answer will be in 16.16 fixed-point, unclipped. |
| * |
| * This function would probably be the prime candidate for SIMD conversion if |
| * you want more speed. |
| */ |
| int32_t fir_convolve(const int16_t* a, const int32_t* b, int num_samples) |
| { |
| int32_t sum = 1 << 13; |
| for (int i = 0; i < num_samples; ++i) { |
| sum += a[i] * (b[i] >> 16); |
| } |
| return sum >> 14; |
| } |
| |
| /* Clip from 16.16 fixed-point to 0.16 fixed-point. */ |
| int16_t clip(int32_t x) |
| { |
| if (x < -32768) { |
| return -32768; |
| } else if (x > 32767) { |
| return 32767; |
| } else { |
| return x; |
| } |
| } |
| |
| /* |
| * Convert a chunk from 44 kHz to 22 kHz. Will update num_samples_in and num_samples_out |
| * accordingly, since it may leave input samples in the buffer due to overlap. |
| * |
| * Input and output are taken to be in 0.16 fixed-point. |
| */ |
| void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out) |
| { |
| if (*num_samples_in < (int)NUM_COEFF_22KHZ) { |
| *num_samples_out = 0; |
| return; |
| } |
| |
| int odd_smp = *num_samples_in & 0x1; |
| int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ; |
| |
| for (int i = 0; i < num_samples; i += 2) { |
| output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ)); |
| } |
| |
| memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input)); |
| *num_samples_out = num_samples / 2; |
| *num_samples_in = OVERLAP_22KHZ + odd_smp; |
| } |
| |
| /* |
| * 2.30 fixed point FIR filter coefficients for conversion 22050 -> 16000, |
| * or 11025 -> 8000. |
| * |
| * Transition band from about 14 kHz, passband ripple < 0.1 dB, |
| * stopband ripple at about -50 dB, linear phase. |
| * |
| * Design and display in MATLAB or Octave using: |
| * |
| * filter = fir1(23, 16000 / 22050); filter = round(filter * 2**30); freqz(filter * 2**-30); |
| */ |
| static const int32_t filter_16khz_coeff[] = { |
| 2057290, -2973608, 1880478, 4362037, |
| -14639744, 18523609, -1609189, -38502470, |
| 78073125, -68353935, -59103896, 617555440, |
| 617555440, -59103896, -68353935, 78073125, |
| -38502470, -1609189, 18523609, -14639744, |
| 4362037, 1880478, -2973608, 2057290, |
| }; |
| #define NUM_COEFF_16KHZ (sizeof(filter_16khz_coeff) / sizeof(filter_16khz_coeff[0])) |
| #define OVERLAP_16KHZ (NUM_COEFF_16KHZ - 1) |
| |
| /* |
| * Convert a chunk from 22 kHz to 16 kHz. Will update num_samples_in and |
| * num_samples_out accordingly, since it may leave input samples in the buffer |
| * due to overlap. |
| * |
| * This implementation is rather ad-hoc; it first low-pass filters the data |
| * into a temporary buffer, and then converts chunks of 441 input samples at a |
| * time into 320 output samples by simple linear interpolation. A better |
| * implementation would use a polyphase filter bank to do these two operations |
| * in one step. |
| * |
| * Input and output are taken to be in 0.16 fixed-point. |
| */ |
| |
| #define RESAMPLE_16KHZ_SAMPLES_IN 441 |
| #define RESAMPLE_16KHZ_SAMPLES_OUT 320 |
| |
| void resample_441_320(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out) |
| { |
| const int num_blocks = (*num_samples_in - OVERLAP_16KHZ) / RESAMPLE_16KHZ_SAMPLES_IN; |
| if (num_blocks < 1) { |
| *num_samples_out = 0; |
| return; |
| } |
| |
| for (int i = 0; i < num_blocks; ++i) { |
| uint32_t tmp[RESAMPLE_16KHZ_SAMPLES_IN]; |
| for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_IN; ++j) { |
| tmp[j] = fir_convolve(input + i * RESAMPLE_16KHZ_SAMPLES_IN + j, |
| filter_16khz_coeff, |
| NUM_COEFF_16KHZ); |
| } |
| |
| const float step_float = (float)RESAMPLE_16KHZ_SAMPLES_IN / (float)RESAMPLE_16KHZ_SAMPLES_OUT; |
| const uint32_t step = (uint32_t)(step_float * 32768.0f + 0.5f); // 17.15 fixed point |
| |
| uint32_t in_sample_num = 0; // 17.15 fixed point |
| for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) { |
| const uint32_t whole = in_sample_num >> 15; |
| const uint32_t frac = (in_sample_num & 0x7fff); // 0.15 fixed point |
| const int32_t s1 = tmp[whole]; |
| const int32_t s2 = tmp[whole + 1]; |
| *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 15)); |
| } |
| |
| } |
| |
| const int samples_consumed = num_blocks * RESAMPLE_16KHZ_SAMPLES_IN; |
| memmove(input, input + samples_consumed, (*num_samples_in - samples_consumed) * sizeof(*input)); |
| *num_samples_in -= samples_consumed; |
| *num_samples_out = RESAMPLE_16KHZ_SAMPLES_OUT * num_blocks; |
| } |
| |
| |
| AudioHardware::DownSampler::DownSampler(uint32_t outSampleRate, |
| uint32_t channelCount, |
| uint32_t frameCount, |
| AudioHardware::BufferProvider* provider) |
| : mStatus(NO_INIT), mProvider(provider), mSampleRate(outSampleRate), |
| mChannelCount(channelCount), mFrameCount(frameCount), |
| mInLeft(NULL), mInRight(NULL), mTmpLeft(NULL), mTmpRight(NULL), |
| mTmp2Left(NULL), mTmp2Right(NULL), mOutLeft(NULL), mOutRight(NULL) |
| |
| { |
| LOGV("AudioHardware::DownSampler() cstor %p SR %d channels %d frames %d", |
| this, mSampleRate, mChannelCount, mFrameCount); |
| |
| if (mSampleRate != 8000 && mSampleRate != 11025 && mSampleRate != 16000 && |
| mSampleRate != 22050) { |
| LOGW("AudioHardware::DownSampler cstor: bad sampling rate: %d", mSampleRate); |
| return; |
| } |
| |
| mInLeft = new int16_t[mFrameCount]; |
| mInRight = new int16_t[mFrameCount]; |
| mTmpLeft = new int16_t[mFrameCount]; |
| mTmpRight = new int16_t[mFrameCount]; |
| mTmp2Left = new int16_t[mFrameCount]; |
| mTmp2Right = new int16_t[mFrameCount]; |
| mOutLeft = new int16_t[mFrameCount]; |
| mOutRight = new int16_t[mFrameCount]; |
| |
| mStatus = NO_ERROR; |
| } |
| |
| AudioHardware::DownSampler::~DownSampler() |
| { |
| if (mInLeft) delete[] mInLeft; |
| if (mInRight) delete[] mInRight; |
| if (mTmpLeft) delete[] mTmpLeft; |
| if (mTmpRight) delete[] mTmpRight; |
| if (mTmp2Left) delete[] mTmp2Left; |
| if (mTmp2Right) delete[] mTmp2Right; |
| if (mOutLeft) delete[] mOutLeft; |
| if (mOutRight) delete[] mOutRight; |
| } |
| |
| void AudioHardware::DownSampler::reset() |
| { |
| mInInBuf = 0; |
| mInTmpBuf = 0; |
| mInTmp2Buf = 0; |
| mOutBufPos = 0; |
| mInOutBuf = 0; |
| } |
| |
| |
| int AudioHardware::DownSampler::resample(int16_t* out, size_t *outFrameCount) |
| { |
| if (mStatus != NO_ERROR) { |
| return mStatus; |
| } |
| |
| if (out == NULL || outFrameCount == NULL) { |
| return BAD_VALUE; |
| } |
| |
| int16_t *outLeft = mTmp2Left; |
| int16_t *outRight = mTmp2Left; |
| if (mSampleRate == 22050) { |
| outLeft = mTmpLeft; |
| outRight = mTmpRight; |
| } else if (mSampleRate == 8000){ |
| outLeft = mOutLeft; |
| outRight = mOutRight; |
| } |
| |
| int outFrames = 0; |
| int remaingFrames = *outFrameCount; |
| |
| if (mInOutBuf) { |
| int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames; |
| |
| for (int i = 0; i < frames; ++i) { |
| out[i] = outLeft[mOutBufPos + i]; |
| } |
| if (mChannelCount == 2) { |
| for (int i = 0; i < frames; ++i) { |
| out[i * 2] = outLeft[mOutBufPos + i]; |
| out[i * 2 + 1] = outRight[mOutBufPos + i]; |
| } |
| } |
| remaingFrames -= frames; |
| mInOutBuf -= frames; |
| mOutBufPos += frames; |
| outFrames += frames; |
| } |
| |
| while (remaingFrames) { |
| LOGW_IF((mInOutBuf != 0), "mInOutBuf should be 0 here"); |
| |
| AudioHardware::BufferProvider::Buffer buf; |
| buf.frameCount = mFrameCount - mInInBuf; |
| int ret = mProvider->getNextBuffer(&buf); |
| if (buf.raw == NULL) { |
| *outFrameCount = outFrames; |
| return ret; |
| } |
| |
| for (size_t i = 0; i < buf.frameCount; ++i) { |
| mInLeft[i + mInInBuf] = buf.i16[i]; |
| } |
| if (mChannelCount == 2) { |
| for (size_t i = 0; i < buf.frameCount; ++i) { |
| mInLeft[i + mInInBuf] = buf.i16[i * 2]; |
| mInRight[i + mInInBuf] = buf.i16[i * 2 + 1]; |
| } |
| } |
| mInInBuf += buf.frameCount; |
| mProvider->releaseBuffer(&buf); |
| |
| /* 44010 -> 22050 */ |
| { |
| int samples_in_left = mInInBuf; |
| int samples_out_left; |
| resample_2_1(mInLeft, mTmpLeft + mInTmpBuf, &samples_in_left, &samples_out_left); |
| |
| if (mChannelCount == 2) { |
| int samples_in_right = mInInBuf; |
| int samples_out_right; |
| resample_2_1(mInRight, mTmpRight + mInTmpBuf, &samples_in_right, &samples_out_right); |
| } |
| |
| mInInBuf = samples_in_left; |
| mInTmpBuf += samples_out_left; |
| mInOutBuf = samples_out_left; |
| } |
| |
| if (mSampleRate == 11025 || mSampleRate == 8000) { |
| /* 22050 - > 11025 */ |
| int samples_in_left = mInTmpBuf; |
| int samples_out_left; |
| resample_2_1(mTmpLeft, mTmp2Left + mInTmp2Buf, &samples_in_left, &samples_out_left); |
| |
| if (mChannelCount == 2) { |
| int samples_in_right = mInTmpBuf; |
| int samples_out_right; |
| resample_2_1(mTmpRight, mTmp2Right + mInTmp2Buf, &samples_in_right, &samples_out_right); |
| } |
| |
| |
| mInTmpBuf = samples_in_left; |
| mInTmp2Buf += samples_out_left; |
| mInOutBuf = samples_out_left; |
| |
| if (mSampleRate == 8000) { |
| /* 11025 -> 8000*/ |
| int samples_in_left = mInTmp2Buf; |
| int samples_out_left; |
| resample_441_320(mTmp2Left, mOutLeft, &samples_in_left, &samples_out_left); |
| |
| if (mChannelCount == 2) { |
| int samples_in_right = mInTmp2Buf; |
| int samples_out_right; |
| resample_441_320(mTmp2Right, mOutRight, &samples_in_right, &samples_out_right); |
| } |
| |
| mInTmp2Buf = samples_in_left; |
| mInOutBuf = samples_out_left; |
| } else { |
| mInTmp2Buf = 0; |
| } |
| |
| } else if (mSampleRate == 16000) { |
| /* 22050 -> 16000*/ |
| int samples_in_left = mInTmpBuf; |
| int samples_out_left; |
| resample_441_320(mTmpLeft, mTmp2Left, &samples_in_left, &samples_out_left); |
| |
| if (mChannelCount == 2) { |
| int samples_in_right = mInTmpBuf; |
| int samples_out_right; |
| resample_441_320(mTmpRight, mTmp2Right, &samples_in_right, &samples_out_right); |
| } |
| |
| mInTmpBuf = samples_in_left; |
| mInOutBuf = samples_out_left; |
| } else { |
| mInTmpBuf = 0; |
| } |
| |
| int frames = (remaingFrames > mInOutBuf) ? mInOutBuf : remaingFrames; |
| |
| for (int i = 0; i < frames; ++i) { |
| out[outFrames + i] = outLeft[i]; |
| } |
| if (mChannelCount == 2) { |
| for (int i = 0; i < frames; ++i) { |
| out[(outFrames + i) * 2] = outLeft[i]; |
| out[(outFrames + i) * 2 + 1] = outRight[i]; |
| } |
| } |
| remaingFrames -= frames; |
| outFrames += frames; |
| mOutBufPos = frames; |
| mInOutBuf -= frames; |
| } |
| |
| return 0; |
| } |
| |
| |
| |
| |
| |
| |
| |
| //------------------------------------------------------------------------------ |
| // Factory |
| //------------------------------------------------------------------------------ |
| |
| extern "C" AudioHardwareInterface* createAudioHardware(void) { |
| return new AudioHardware(); |
| } |
| |
| }; // namespace android |
