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Diffstat (limited to 'libaudio/AudioHardwareALSA.cpp')
| -rwxr-xr-x | libaudio/AudioHardwareALSA.cpp | 2654 |
1 files changed, 0 insertions, 2654 deletions
diff --git a/libaudio/AudioHardwareALSA.cpp b/libaudio/AudioHardwareALSA.cpp deleted file mode 100755 index 45223f3..0000000 --- a/libaudio/AudioHardwareALSA.cpp +++ /dev/null @@ -1,2654 +0,0 @@ -/* AudioHardwareALSA.cpp - ** - ** Copyright 2008-2009 Wind River Systems - ** - ** 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 <errno.h> -#include <stdarg.h> -#include <stdint.h> -#include <sys/types.h> -#include <sys/stat.h> -#include <fcntl.h> -#include <stdlib.h> -#include <unistd.h> - -//#define LOG_NDEBUG 0 -#define LOG_TAG "AudioHardwareALSA" -#include <utils/Log.h> -#include <utils/String8.h> - -#include <cutils/properties.h> -#include <media/AudioRecord.h> -#include <hardware_legacy/power.h> - -#include <alsa/asoundlib.h> -#include "AudioHardwareALSA.h" -// #define READ_FRAME_SIZE 2080 -// #define READ_FRAME_SIZE_STANDARD 4160 - -#include <dlfcn.h> - -#define SND_MIXER_VOL_RANGE_MIN (0) -#define SND_MIXER_VOL_RANGE_MAX (100) - -#define ALSA_NAME_MAX 128 - -#define ALSA_STRCAT(x,y) \ - if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \ - strcat(x, y); - -extern "C" -{ - extern int ffs(int i); - - // - // Make sure this prototype is consistent with what's in - // external/libasound/alsa-lib-1.0.16/src/pcm/pcm_null.c! - // - extern int snd_pcm_null_open(snd_pcm_t **pcmp, - const char *name, - snd_pcm_stream_t stream, - int mode); - - // - // Function for dlsym() to look up for creating a new AudioHardwareInterface. - // - android::AudioHardwareInterface *createAudioHardware(void) { - return new android::AudioHardwareALSA(); - } -} // extern "C" - -namespace android -{ - -typedef AudioSystem::audio_devices audio_routes; -#define ROUTE_ALL AudioSystem::DEVICE_OUT_ALL -#define ROUTE_EARPIECE AudioSystem::DEVICE_OUT_EARPIECE -#define ROUTE_HEADSET AudioSystem::DEVICE_OUT_WIRED_HEADSET -#define ROUTE_HEADPHONE AudioSystem::DEVICE_OUT_WIRED_HEADPHONE -#define ROUTE_SPEAKER AudioSystem::DEVICE_OUT_SPEAKER -#define ROUTE_BLUETOOTH_SCO AudioSystem::DEVICE_OUT_BLUETOOTH_SCO -#define ROUTE_BLUETOOTH_SCO_HEADSET AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET -#define ROUTE_BLUETOOTH_SCO_CARKIT AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT -#define ROUTE_BLUETOOTH_A2DP AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP -#define ROUTE_BLUETOOTH_A2DP_HEADPHONES AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES -#define ROUTE_BLUETOOTH_A2DP_SPEAKER AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER - -// ---------------------------------------------------------------------------- - - -static const char _nullALSADeviceName[] = "NULL_Device"; - -static void ALSAErrorHandler(const char *file, - int line, - const char *function, - int err, - const char *fmt, - ...) -{ - char buf[BUFSIZ]; - va_list arg; - int l; - - va_start(arg, fmt); - l = snprintf(buf, BUFSIZ, "%s:%i:(%s) ", file, line, function); - vsnprintf(buf + l, BUFSIZ - l, fmt, arg); - buf[BUFSIZ-1] = '\0'; - LOGE("ALSALib %s.", buf); - va_end(arg); -} - -// ---------------------------------------------------------------------------- - -/* The following table(s) need to match in order of the route bits - */ -static const char *deviceSuffix[] = { - // output devices - /* ROUTE_EARPIECE */ "_Earpiece", - /* ROUTE_SPEAKER */ "_Speaker", - /* ROUTE_HEADSET */ "_Headset", - /* ROUTE_HEADPHONE */ "_Headphone", - /* ROUTE_BLUETOOTH_SCO */ "_Bluetooth", - /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", - /* ROUTE_BLUETOOTH_SCO_CARKIT */ "_Bluetooth", //"_Bluetooth_Carkit" - /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth", //"_Bluetooth-A2DP" - /* ROUTE_BLUETOOTH_A2DP_HEADPHONES */ "_Bluetooth", //"_Bluetooth-A2DP_HeadPhone" - /* ROUTE_BLUETOOTH_A2DP_SPEAKER */ "_Bluetooth", // "_Bluetooth-A2DP_Speaker" - /* ROUTE_AUX_DIGITAL */ "_AuxDigital", - /* ROUTE_TV_OUT */ "_TvOut", - /* ROUTE_AUX_DIGITAL */ "_ExtraDockSpeaker", - /* ROUTE_NULL */ "_Null", - /* ROUTE_NULL */ "_Null", - /* ROUTE_DEFAULT */ "_OutDefault", - - // input devices - /* ROUTE_COMMUNICATION */ "_Communication", - /* ROUTE_AMBIENT */ "_Ambient", - /* ROUTE_BUILTIN_MIC */ "_Speaker", - /* ROUTE_BLUETOOTH_SCO_HEADSET */ "_Bluetooth", - /* ROUTE_WIRED_HEADSET */ "_Headset", - /* ROUTE_AUX_DIGITAL */ "_AuxDigital", - /* ROUTE_VOICE_CALL */ "_VoiceCall", - /* ROUTE_BACK_MIC */ "_BackMic", - /* ROUTE_IN_DEFAULT */ "_InDefault", -}; - -static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *)); - -struct mixer_info_t; - -struct alsa_properties_t -{ - const audio_routes routes; - const char *propName; - const char *propDefault; - mixer_info_t *mInfo; -}; - -static alsa_properties_t masterPlaybackProp = { - ROUTE_ALL, "alsa.mixer.playback.master", "PCM", NULL -}; - -static alsa_properties_t masterCaptureProp = { - ROUTE_ALL, "alsa.mixer.capture.master", "Capture", NULL -}; - -static alsa_properties_t -mixerMasterProp[SND_PCM_STREAM_LAST+1] = { - { ROUTE_ALL, "alsa.mixer.playback.master", "PCM", NULL}, - { ROUTE_ALL, "alsa.mixer.capture.master", "Capture", NULL} -}; - -static alsa_properties_t -mixerProp[][SND_PCM_STREAM_LAST+1] = { - { - {ROUTE_EARPIECE, "alsa.mixer.playback.earpiece", "Earpiece", NULL}, - {ROUTE_EARPIECE, "alsa.mixer.capture.earpiece", "Capture", NULL} - }, - { - {ROUTE_SPEAKER, "alsa.mixer.playback.speaker", "Speaker", NULL}, - {ROUTE_SPEAKER, "alsa.mixer.capture.speaker", "", NULL} - }, - { - {ROUTE_BLUETOOTH_SCO, "alsa.mixer.playback.bluetooth.sco", "Bluetooth", NULL}, - {ROUTE_BLUETOOTH_SCO, "alsa.mixer.capture.bluetooth.sco", "Bluetooth Capture", NULL} - }, - { - {ROUTE_HEADSET, "alsa.mixer.playback.headset", "Headphone", NULL}, - {ROUTE_HEADSET, "alsa.mixer.capture.headset", "Capture", NULL} - }, - { - {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.playback.bluetooth.a2dp", "Bluetooth A2DP", NULL}, - {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.capture.bluetooth.a2dp", "Bluetooth A2DP Capture", NULL} - }, - { - {static_cast<audio_routes>(0), NULL, NULL, NULL}, - {static_cast<audio_routes>(0), NULL, NULL, NULL} - } -}; - -const uint32_t AudioHardwareALSA::inputSamplingRates[] = { - 8000, 11025, 16000, 22050, 44100 -}; - -// ---------------------------------------------------------------------------- - -AudioHardwareALSA::AudioHardwareALSA() : - mOutput(0), - mInput(0), - mSecRilLibHandle(NULL), - mRilClient(0), - mVrModeEnabled(false), - mActivatedCP(false), - mBluetoothECOff(false) -{ - snd_lib_error_set_handler(&ALSAErrorHandler); - mMixer = new ALSAMixer; - - loadRILD(); -} - -AudioHardwareALSA::~AudioHardwareALSA() -{ - if (mOutput) delete mOutput; - if (mInput) delete mInput; - if (mMixer) delete mMixer; - - 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; - } -} - - -void AudioHardwareALSA::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 AudioHardwareALSA::initCheck() -{ - if (mMixer && mMixer->isValid()) - return NO_ERROR; - else - return NO_INIT; -} - - -status_t AudioHardwareALSA::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; -} - - -status_t AudioHardwareALSA::setVoiceVolume(float volume) -{ - LOGI("### setVoiceVolume"); - - AutoMutex lock(mLock); - // sangsu fix : transmic volume level IPC to modem - if ( (AudioSystem::MODE_IN_CALL == mMode) && (mSecRilLibHandle) && - (connectRILDIfRequired() == OK) ) { - - uint32_t routes = AudioSystem::ROUTE_EARPIECE; - if (mOutput != NULL) { - routes = mOutput->device(); - } - int int_volume = (int)(volume * 5); - - LOGI("### route(%d) call volume(%f)", routes, volume); - switch (routes) { - case AudioSystem::ROUTE_EARPIECE: - LOGI("### earpiece call volume"); - setCallVolume(mRilClient, SOUND_TYPE_VOICE, int_volume); - break; - - case AudioSystem::ROUTE_SPEAKER: - LOGI("### speaker call volume"); - setCallVolume(mRilClient, SOUND_TYPE_SPEAKER, int_volume); - break; - - case AudioSystem::ROUTE_BLUETOOTH_SCO: - case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: - case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - case AudioSystem::ROUTE_BLUETOOTH_A2DP: - LOGI("### bluetooth call volume"); - setCallVolume(mRilClient, SOUND_TYPE_BTVOICE, int_volume); - break; - - case AudioSystem::ROUTE_HEADSET: - case AudioSystem::ROUTE_HEADPHONE: // Use receive path with 3 pole headset. - LOGI("### headset call volume"); - setCallVolume(mRilClient, SOUND_TYPE_HEADSET, int_volume); - break; - - default: - LOGE("### Call volume setting error!!!0x%08x \n", routes); - break; - } - } - // sangsu fix end - - // The voice volume is used by the VOICE_CALL audio stream. - if (mMixer) - return mMixer->setVolume(ROUTE_EARPIECE, volume); - else - return INVALID_OPERATION; -} - -status_t AudioHardwareALSA::setMasterVolume(float volume) -{ - if (mMixer) - return mMixer->setMasterVolume(volume); - else - return INVALID_OPERATION; -} - -AudioStreamOut * -AudioHardwareALSA::openOutputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status) -{ - AudioStreamOutALSA *out = NULL; - status_t ret = NO_ERROR; - { - AutoMutex lock(mLock); - - // only one output stream allowed - if (mOutput) { - ret = ALREADY_EXISTS; - goto exit; - } - - LOGV("[[[[[[[[\n%s - format = %d, channels = %d, sampleRate = %d, devices = %d]]]]]]]]\n", __func__, *format, *channels, *sampleRate,devices); - - out = new AudioStreamOutALSA(this); - - ret = out->set(format, channels, sampleRate); - - if (ret == NO_ERROR) { - mOutput = out; - } - } -exit: - if (ret == NO_ERROR) { - // Some information is expected to be available immediately after - // the device is open. - /* Tushar - Sets the current device output here - we may set device here */ - LOGI("%s] Setting ALSA device.", __func__); - mOutput->setDevice(mMode, devices, PLAYBACK); /* tushar - Enable all devices as of now */ - } else if (out) { - delete out; - } - if (status) { - *status = ret; - } - return mOutput; -} - -void -AudioHardwareALSA::closeOutputStream(AudioStreamOut* out) -{ - /* TODO:Tushar: May lead to segmentation fault - check*/ - //delete out; - { - AutoMutex lock(mLock); - - if (mOutput == 0 || mOutput != out) { - LOGW("Attempt to close invalid output stream"); - return; - } - mOutput = 0; - } - delete out; -} - - -AudioStreamIn* -AudioHardwareALSA::openInputStream( - uint32_t devices, - int *format, - uint32_t *channels, - uint32_t *sampleRate, - status_t *status, - AudioSystem::audio_in_acoustics acoustics) -{ - AudioStreamInALSA *in = NULL; - status_t ret = NO_ERROR; - { - AutoMutex lock(mLock); - - // only one input stream allowed - if (mInput) { - ret = ALREADY_EXISTS; - goto exit; - } - - in = new AudioStreamInALSA(this); - - ret = in->set(format, channels, sampleRate); - if (ret == NO_ERROR) { - mInput = in; - } - } -exit: - if (ret == NO_ERROR) { - // Some information is expected to be available immediately after - // the device is open. - mInput->setDevice(mMode, devices, CAPTURE); /* Tushar - as per modified arch */ - } else if (in != NULL) { - delete in; - } - if (status) { - *status = ret; - } - return mInput; -} - -void -AudioHardwareALSA::closeInputStream(AudioStreamIn* in) -{ - /* TODO:Tushar: May lead to segmentation fault - check*/ - //delete in; - { - AutoMutex lock(mLock); - - if (mInput == 0 || mInput != in) { - LOGW("Attempt to close invalid input stream"); - return; - } else { - mInput = 0; - } - } - delete in; -} - - -status_t AudioHardwareALSA::doRouting(uint32_t device, bool force) -{ - AutoMutex lock(mLock); - return doRouting_l(device, force); -} - -status_t AudioHardwareALSA::doRouting_l(uint32_t device, bool force) -{ - status_t ret; - int mode = mMode; // Prevent to changing mode on setup sequence. - - LOGV("doRouting: device %x, force %d", device, force); - - if (mOutput) { - //device = 0; /* Tushar - temp implementation */ - if (device == AudioSystem::DEVICE_OUT_DEFAULT) { - device = mOutput->device(); - } - - // Setup sound path for CP clocking - if ( (AudioSystem::MODE_IN_CALL == mode) && (mSecRilLibHandle) && - (connectRILDIfRequired() == OK) ) { - - LOGI("### incall mode route (%d)", device); - - switch(device){ - case AudioSystem::ROUTE_EARPIECE: - LOGI("### incall mode earpiece route"); - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HANDSET); - break; - - case AudioSystem::ROUTE_SPEAKER: - LOGI("### incall mode speaker route"); - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_SPEAKER); - break; - - case AudioSystem::ROUTE_BLUETOOTH_SCO: - case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET: - case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT: - LOGI("### incall mode bluetooth route %s NR", mBluetoothECOff ? "NO" : ""); - if (mBluetoothECOff) - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH_NO_NR); - else - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH); - break; - - case AudioSystem::ROUTE_HEADSET : - case AudioSystem::ROUTE_HEADPHONE : - LOGI("### incall mode headset route"); - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_HEADSET); - break; - - case AudioSystem::ROUTE_BLUETOOTH_A2DP: - LOGI("### incall mode bluetooth route"); - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH); - break; - - default: - LOGE("### incall mode Error!! route = [%d]", device); - break; - } - } - - ret = mOutput->setDevice(mode, device, PLAYBACK, force); - - return ret; - } - - return NO_INIT; -} - - -status_t AudioHardwareALSA::setMicMute(bool state) -{ - if (mMixer) - return mMixer->setCaptureMuteState(ROUTE_EARPIECE, state); - - return NO_INIT; -} - -status_t AudioHardwareALSA::getMicMute(bool *state) -{ - if (mMixer) - return mMixer->getCaptureMuteState(ROUTE_EARPIECE, state); - - return NO_ERROR; -} - -status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args) -{ - return NO_ERROR; -} - - -uint32_t AudioHardwareALSA::bufferRatio(uint32_t samplingRate) { - switch (samplingRate) { - case 8000: - case 11025: - return 4; - case 16000: - case 22050: - return 2; - case 44100: - default: - break; - } - return 1; -} - - -size_t AudioHardwareALSA::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) -{ - if (sampleRate != 8000 && sampleRate != 11025 && sampleRate != 16000 && - sampleRate != 22050 && sampleRate != 44100) { - LOGW("getInputBufferSize bad sampling rate: %d", sampleRate); - return 0; - } - if (format != AudioSystem::PCM_16_BIT) { - LOGW("getInputBufferSize bad format: %d", format); - return 0; - } - if (channelCount != 1) { - LOGW("getInputBufferSize bad channel count: %d", channelCount); - return 0; - } - - size_t size = (PERIOD_SZ_CAPTURE / bufferRatio(sampleRate)) * sizeof(int16_t); - LOGV("getInputBufferSize() rate %d, ratio %d", sampleRate, size); - return size; - -} - -uint32_t AudioHardwareALSA::checkInputSampleRate(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]; -} - -status_t AudioHardwareALSA::setMode(int mode) -{ - AutoMutex lock(mLock); - int prevMode = mMode; - status_t status = AudioHardwareBase::setMode(mode); - LOGV("setMode() : new %d, old %d", mMode, prevMode); - if (status == NO_ERROR) { - if ( (mMode == AudioSystem::MODE_RINGTONE) || (mMode == AudioSystem::MODE_IN_CALL) ) - { - if ( (!mActivatedCP) && (mSecRilLibHandle) && (connectRILDIfRequired() == OK) ) { - setCallClockSync(mRilClient, SOUND_CLOCK_START); - mActivatedCP = true; - } - } - - // make sure that doAudioRouteOrMute() is called by doRouting() - // when entering or exiting in call mode even if the new device - // selected is the same as current one. - if ((prevMode != AudioSystem::MODE_IN_CALL) && (mMode == AudioSystem::MODE_IN_CALL)) { - LOGV("setMode() entering call"); - doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true); - setVoiceRecordGain_l(false); - } - if ((prevMode == AudioSystem::MODE_IN_CALL) && (mMode != AudioSystem::MODE_IN_CALL)) { - LOGV("setMode() exiting call"); - doRouting_l(AudioSystem::DEVICE_OUT_DEFAULT, true); - if (mOutput != NULL && !mOutput->isActive()) { - mOutput->close(); - } - } - - if (mMode == AudioSystem::MODE_NORMAL) { - if(mActivatedCP) - mActivatedCP = false; - } - } - - return status; -} - -int AudioHardwareALSA::setVoiceRecordGain(bool enable) -{ - AutoMutex lock(mLock); - return setVoiceRecordGain_l(enable); -} - -int AudioHardwareALSA::setVoiceRecordGain_l(bool enable) -{ - LOGI("[%s], enable=%d", __func__, enable); - if (enable != mVrModeEnabled && - !(enable && (mMode == AudioSystem::MODE_IN_CALL))) { - ALSAControl *alsaControl = new ALSAControl(); - status_t ret = alsaControl->set("Recognition Control", enable ? RECOGNITION_ON : RECOGNITION_OFF); - delete alsaControl; - mVrModeEnabled = enable; - } - - return NO_ERROR; -} - -status_t AudioHardwareALSA::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 bt_nrec_key = String8("bt_headset_nrec"); - String8 value; - - LOGV("setParameters(%s)", keyValuePairs.string()); - - if (param.get(bt_nrec_key, value) == NO_ERROR) { - setBluetoothNrEcOnOff((value == "on") ? false : true); - } - - return NO_ERROR; -} - -void AudioHardwareALSA::setBluetoothNrEcOnOff(bool disable) -{ - LOGV("setBluetoothNrEcOnOff(%s)", disable ? "true" : "false"); - - if (disable != mBluetoothECOff) - { - mBluetoothECOff = disable; - - if ( (mOutput) && (AudioSystem::MODE_IN_CALL == mMode) && - (mSecRilLibHandle) && (connectRILDIfRequired() == OK)) { - - uint32_t device = mOutput->device(); - - switch (device) { - case AudioSystem::ROUTE_BLUETOOTH_SCO: - case AudioSystem::ROUTE_BLUETOOTH_SCO_HEADSET: - case AudioSystem::ROUTE_BLUETOOTH_SCO_CARKIT: - LOGV("### incall mode bluetooth EC %s route", mBluetoothECOff ? "OFF" : "ON"); - if (mBluetoothECOff) - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH_NO_NR); - else - setCallAudioPath(mRilClient, SOUND_AUDIO_PATH_BLUETOOTH); - break; - - default : - LOGE("Bluetooth path is not activated!!"); - break; - } - } - } -} - - -// ---------------------------------------------------------------------------- - -ALSAStreamOps::ALSAStreamOps() : - mHandle(0), - mHardwareParams(0), - mSoftwareParams(0), - mDevice(0) -{ - if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) { - LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!"); - } - - if (snd_pcm_sw_params_malloc(&mSoftwareParams) < 0) { - LOG_ALWAYS_FATAL("Failed to allocate ALSA software parameters!"); - } -} - -ALSAStreamOps::~ALSAStreamOps() -{ - AutoMutex lock(mLock); - - close(); - - if (mHardwareParams) - snd_pcm_hw_params_free(mHardwareParams); - - if (mSoftwareParams) - snd_pcm_sw_params_free(mSoftwareParams); -} - - -status_t ALSAStreamOps::set(int *pformat, - uint32_t *pchannels, - uint32_t *prate) -{ - int lformat = pformat ? *pformat : 0; - unsigned int lchannels = pchannels ? *pchannels : 0; - unsigned int lrate = prate ? *prate : 0; - - - LOGD("ALSAStreamOps - input - format = %d, channels = %d, rate = %d\n", lformat, lchannels, lrate); - LOGD("ALSAStreamOps - default - format = %d, channelCount = %d, rate = %d\n", mDefaults->format, mDefaults->channelCount, mDefaults->sampleRate); - - if (lformat == 0) lformat = getAndroidFormat(mDefaults->format);//format(); - if (lchannels == 0) lchannels = getAndroidChannels(mDefaults->channelCount);// channelCount(); - if (lrate == 0) lrate = mDefaults->sampleRate; - - if ( (lformat != getAndroidFormat(mDefaults->format)) || - (lchannels != getAndroidChannels(mDefaults->channelCount)) ) { - if (pformat) *pformat = getAndroidFormat(mDefaults->format); - if (pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); - return BAD_VALUE; - } - if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { - if (lrate != mDefaults->sampleRate) { - if (prate) *prate = mDefaults->sampleRate; - return BAD_VALUE; - } - } else { - uint32_t rate = AudioHardwareALSA::checkInputSampleRate(lrate); - if (rate != lrate) { - if (prate) *prate = rate; - return BAD_VALUE; - } - lrate = rate; - } - mDefaults->bufferRatio = AudioHardwareALSA::bufferRatio(lrate); - mDefaults->sampleRate = lrate; - - if(pformat) *pformat = getAndroidFormat(mDefaults->format); - if(pchannels) *pchannels = getAndroidChannels(mDefaults->channelCount); - if(prate) *prate = mDefaults->sampleRate; - - return NO_ERROR; -} - - -uint32_t ALSAStreamOps::sampleRate() const -{ - return mDefaults->sampleRate; -} - -status_t ALSAStreamOps::sampleRate(uint32_t rate) -{ - const char *stream; - unsigned int requestedRate; - int err; - - if (!mHandle) - return NO_INIT; - - stream = streamName(); - requestedRate = rate; - err = snd_pcm_hw_params_set_rate_near(mHandle, - mHardwareParams, - &requestedRate, - 0); - - if (err < 0) { - LOGE("Unable to set %s sample rate to %u: %s", - stream, rate, snd_strerror(err)); - return BAD_VALUE; - } - if (requestedRate != rate) { - // Some devices have a fixed sample rate, and can not be changed. - // This may cause resampling problems; i.e. PCM playback will be too - // slow or fast. - LOGW("Requested rate (%u HZ) does not match actual rate (%u HZ)", - rate, requestedRate); - } - else { - LOGD("Set %s sample rate to %u HZ", stream, requestedRate); - } - return NO_ERROR; -} - -// -// Return the number of bytes (not frames) -// -size_t ALSAStreamOps::bufferSize() const -{ - int err; - - size_t size = ((mDefaults->periodSize / mDefaults->bufferRatio) * mDefaults->channelCount * - snd_pcm_format_physical_width(mDefaults->format)) / 8; - LOGV("bufferSize() channelCount %d, bufferRatio %d, size %d", - mDefaults->channelCount, mDefaults->bufferRatio, size); - return size; - -} - -int ALSAStreamOps::getAndroidFormat(snd_pcm_format_t format) -{ - int pcmFormatBitWidth; - int audioSystemFormat; - - pcmFormatBitWidth = snd_pcm_format_physical_width(format); - audioSystemFormat = AudioSystem::DEFAULT; - switch(pcmFormatBitWidth) { - case 8: - audioSystemFormat = AudioSystem::PCM_8_BIT; - break; - - case 16: - audioSystemFormat = AudioSystem::PCM_16_BIT; - break; - - default: - LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth); - } - - return audioSystemFormat; - -} - -int ALSAStreamOps::format() const -{ - snd_pcm_format_t ALSAFormat; - int pcmFormatBitWidth; - int audioSystemFormat; - - if (snd_pcm_hw_params_get_format(mHardwareParams, &ALSAFormat) < 0) { - return -1; - } - - pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat); - audioSystemFormat = AudioSystem::DEFAULT; - switch(pcmFormatBitWidth) { - case 8: - audioSystemFormat = AudioSystem::PCM_8_BIT; - break; - - case 16: - audioSystemFormat = AudioSystem::PCM_16_BIT; - break; - - default: - LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth); - } - - return audioSystemFormat; -} - -uint32_t ALSAStreamOps::getAndroidChannels(int channelCount) const -{ - int AudioSystemChannels = AudioSystem::DEFAULT; - - if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { - switch(channelCount){ - case 1: - AudioSystemChannels = AudioSystem::CHANNEL_OUT_MONO; - break; - case 2: - AudioSystemChannels = AudioSystem::CHANNEL_OUT_STEREO; - break; - case 4: - AudioSystemChannels = AudioSystem::CHANNEL_OUT_QUAD; - break; - case 6: - AudioSystemChannels = AudioSystem::CHANNEL_OUT_5POINT1; - break; - default: - LOGE("FATAL: AudioSystem does not support %d output channels.", channelCount); - } - } else { - switch(channelCount){ - case 1: - AudioSystemChannels = AudioSystem::CHANNEL_IN_MONO; - break; - case 2: - AudioSystemChannels = AudioSystem::CHANNEL_IN_STEREO; - break; - default: - LOGE("FATAL: AudioSystem does not support %d input channels.", channelCount); - } - - } - return AudioSystemChannels; -} - -uint32_t ALSAStreamOps::channels() const -{ - return getAndroidChannels(mDefaults->channelCount); -} - -int ALSAStreamOps::channelCount() const -{ - return mDefaults->channelCount; -} - -status_t ALSAStreamOps::channelCount(int channelCount) { - int err; - - if (!mHandle) - return NO_INIT; - - err = snd_pcm_hw_params_set_channels(mHandle, mHardwareParams, channelCount); - if (err < 0) { - LOGE("Unable to set channel count to %i: %s", - channelCount, snd_strerror(err)); - return BAD_VALUE; - } - - LOGD("Using %i %s for %s.", - channelCount, channelCount == 1 ? "channel" : "channels", streamName()); - - return NO_ERROR; -} - -status_t ALSAStreamOps::open(int mode, uint32_t device) -{ - const char *stream = streamName(); - const char *devName = deviceName(mode, device); - - int err; - - LOGI("Try to open ALSA %s device %s", stream, devName); - - for(;;) { - // The PCM stream is opened in blocking mode, per ALSA defaults. The - // AudioFlinger seems to assume blocking mode too, so asynchronous mode - // should not be used. - err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0); - if (err == 0) break; - - // See if there is a less specific name we can try. - // Note: We are changing the contents of a const char * here. - char *tail = strrchr(devName, '_'); - if (! tail) break; - *tail = 0; - } - - if (err < 0) { - // None of the Android defined audio devices exist. Open a generic one. - devName = "hw:00,1"; // 090507 SMDKC110 Froyo - - err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0); - if (err < 0) { - // Last resort is the NULL device (i.e. the bit bucket). - devName = _nullALSADeviceName; - err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0); - } - } - - mDevice = device; - - LOGI("Initialized ALSA %s device %s", stream, devName); - return err; -} - -void ALSAStreamOps::close() -{ - snd_pcm_t *handle = mHandle; - mHandle = NULL; - - if (handle) { - LOGV("ALSAStreamOps::close()"); - snd_pcm_drain(handle); - snd_pcm_close(handle); - } -} - -status_t ALSAStreamOps::setSoftwareParams() -{ - if (!mHandle) - return NO_INIT; - - int err; - - // Get the current software parameters - err = snd_pcm_sw_params_current(mHandle, mSoftwareParams); - if (err < 0) { - LOGE("Unable to get software parameters: %s", snd_strerror(err)); - return NO_INIT; - } - - snd_pcm_uframes_t bufferSize = 0; - snd_pcm_uframes_t periodSize = 0; - snd_pcm_uframes_t startThreshold; - - // Configure ALSA to start the transfer when the buffer is almost full. - snd_pcm_get_params(mHandle, &bufferSize, &periodSize); - LOGE("bufferSize %d, periodSize %d\n", (int)bufferSize, (int)periodSize); - - if (mDefaults->direction == SND_PCM_STREAM_PLAYBACK) { - // For playback, configure ALSA to start the transfer when the - // buffer is almost full. - startThreshold = (bufferSize / periodSize) * periodSize; - //startThreshold = 1; - } - else { - // For recording, configure ALSA to start the transfer on the - // first frame. - startThreshold = 1; - } - - err = snd_pcm_sw_params_set_start_threshold(mHandle, - mSoftwareParams, - startThreshold); - if (err < 0) { - LOGE("Unable to set start threshold to %lu frames: %s", - startThreshold, snd_strerror(err)); - return NO_INIT; - } - - // Stop the transfer when the buffer is full. - err = snd_pcm_sw_params_set_stop_threshold(mHandle, - mSoftwareParams, - bufferSize); - if (err < 0) { - LOGE("Unable to set stop threshold to %lu frames: %s", - bufferSize, snd_strerror(err)); - return NO_INIT; - } - - // Allow the transfer to start when at least periodSize samples can be - // processed. - err = snd_pcm_sw_params_set_avail_min(mHandle, - mSoftwareParams, - periodSize); - if (err < 0) { - LOGE("Unable to configure available minimum to %lu: %s", - periodSize, snd_strerror(err)); - return NO_INIT; - } - - // Commit the software parameters back to the device. - err = snd_pcm_sw_params(mHandle, mSoftwareParams); - if (err < 0) { - LOGE("Unable to configure software parameters: %s", - snd_strerror(err)); - return NO_INIT; - } - - return NO_ERROR; -} - -status_t ALSAStreamOps::setPCMFormat(snd_pcm_format_t format) -{ - const char *formatDesc; - const char *formatName; - bool validFormat; - int err; - - // snd_pcm_format_description() and snd_pcm_format_name() do not perform - // proper bounds checking. - validFormat = (static_cast<int>(format) > SND_PCM_FORMAT_UNKNOWN) && - (static_cast<int>(format) <= SND_PCM_FORMAT_LAST); - formatDesc = validFormat ? - snd_pcm_format_description(format) : "Invalid Format"; - formatName = validFormat ? - snd_pcm_format_name(format) : "UNKNOWN"; - - err = snd_pcm_hw_params_set_format(mHandle, mHardwareParams, format); - if (err < 0) { - LOGE("Unable to configure PCM format %s (%s): %s", - formatName, formatDesc, snd_strerror(err)); - return NO_INIT; - } - - LOGD("Set %s PCM format to %s (%s)", streamName(), formatName, formatDesc); - return NO_ERROR; -} - -status_t ALSAStreamOps::setHardwareResample(bool resample) -{ - int err; - - err = snd_pcm_hw_params_set_rate_resample(mHandle, - mHardwareParams, - static_cast<int>(resample)); - if (err < 0) { - LOGE("Unable to %s hardware resampling: %s", - resample ? "enable" : "disable", - snd_strerror(err)); - return NO_INIT; - } - return NO_ERROR; -} - -const char *ALSAStreamOps::streamName() -{ - // Don't use snd_pcm_stream(mHandle), as the PCM stream may not be - // opened yet. In such case, snd_pcm_stream() will abort(). - return snd_pcm_stream_name(mDefaults->direction); -} - -// -// Set playback or capture PCM device. It's possible to support audio output -// or input from multiple devices by using the ALSA plugins, but this is -// not supported for simplicity. -// -// The AudioHardwareALSA API does not allow one to set the input routing. -// -// If the "routes" value does not map to a valid device, the default playback -// device is used. -// -status_t ALSAStreamOps::setDevice(int mode, uint32_t device, uint audio_mode) -{ - // Close off previously opened device. - // It would be nice to determine if the underlying device actually - // changes, but we might be manipulating mixer settings (see asound.conf). - // - close(); - - const char *stream = streamName(); - - - LOGD("\n------------------------>>>>>> ALSA OPEN mode %d,device %d \n",mode,device); - - status_t status = open (mode, device); - int err; - unsigned int period_val; - - if (status != NO_ERROR) - return status; - - err = snd_pcm_hw_params_any(mHandle, mHardwareParams); - if (err < 0) { - LOGE("Unable to configure hardware: %s", snd_strerror(err)); - return NO_INIT; - } - - status = setPCMFormat(mDefaults->format); - - // Set the interleaved read and write format. - err = snd_pcm_hw_params_set_access(mHandle, mHardwareParams, - SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) { - LOGE("Unable to configure PCM read/write format: %s", - snd_strerror(err)); - return NO_INIT; - } - - - // - // Some devices do not have the default two channels. Force an error to - // prevent AudioMixer from crashing and taking the whole system down. - // - // Note that some devices will return an -EINVAL if the channel count - // is queried before it has been set. i.e. calling channelCount() - // before channelCount(channels) may return -EINVAL. - // - status = channelCount(mDefaults->channelCount); - if (status != NO_ERROR) - return status; - - // Don't check for failure; some devices do not support the default - // sample rate. - // FIXME:: always use default sampling rate - sampleRate(DEFAULT_SAMPLE_RATE); - - snd_pcm_uframes_t bufferSize = mDefaults->bufferSize; - snd_pcm_uframes_t periodSize = mDefaults->periodSize; - period_val = bufferSize/periodSize; - - unsigned int latency = mDefaults->latency; - - // Make sure we have at least the size we originally wanted - err = snd_pcm_hw_params_set_buffer_size(mHandle, mHardwareParams, bufferSize); - if (err < 0) { - LOGE("Unable to set buffer size to %d: %s", - (int)bufferSize, snd_strerror(err)); - return NO_INIT; - } - - err = snd_pcm_hw_params_set_period_size (mHandle, mHardwareParams, periodSize, NULL); - if (err < 0) { - LOGE("Unable to set the period size for latency: %s", snd_strerror(err)); - return NO_INIT; - } - - err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL); - if (err < 0) { - LOGE("Unable to get the period size for latency: %s", snd_strerror(err)); - return NO_INIT; - } - -// err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL); -// if (err < 0) { -// LOGE("Unable to get the period size for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } - -// // Setup buffers for latency -// err = snd_pcm_hw_params_set_buffer_time_near (mHandle, mHardwareParams, -// &latency, NULL); -// if(audio_mode == PLAYBACK) { -// period_val = PERIODS_PLAYBACK; -// if(snd_pcm_hw_params_set_periods(mHandle, mHardwareParams, period_val, 0) < 0) -// LOGE("Fail to set period size %d for playback", period_val); -// } -// else -// period_val = PERIODS_CAPTURE; -// -// if (err < 0) { -// LOGD("snd_pcm_hw_params_set_buffer_time_near() failed: %s", snd_strerror(err)); -// /* That didn't work, set the period instead */ -// unsigned int periodTime = latency / period_val; -// err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams, -// &periodTime, NULL); -// if (err < 0) { -// LOGE("Unable to set the period time for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL); -// if (err < 0) { -// LOGE("Unable to get the period size for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// bufferSize = periodSize * period_val; -// if (bufferSize < mDefaults->bufferSize) -// bufferSize = mDefaults->bufferSize; -// err = snd_pcm_hw_params_set_buffer_size_near (mHandle, mHardwareParams, &bufferSize); -// if (err < 0) { -// LOGE("Unable to set the buffer size for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// } else { -// LOGD("snd_pcm_hw_params_set_buffer_time_near() OK"); -// // OK, we got buffer time near what we expect. See what that did for bufferSize. -// err = snd_pcm_hw_params_get_buffer_size (mHardwareParams, &bufferSize); -// if (err < 0) { -// LOGE("Unable to get the buffer size for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// // Does set_buffer_time_near change the passed value? It should. -// err = snd_pcm_hw_params_get_buffer_time (mHardwareParams, &latency, NULL); -// if (err < 0) { -// LOGE("Unable to get the buffer time for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// LOGD("got latency %d for bufferSize %d", latency, bufferSize); -// unsigned int periodTime = latency / period_val; -// LOGD("got latency %d for bufferSize %d => periodTime %d", latency, bufferSize, periodTime); -// err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams, -// &periodTime, NULL); -// if (err < 0) { -// LOGE("Unable to set the period time for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL); -// if (err < 0) { -// LOGE("Unable to get the period size for latency: %s", snd_strerror(err)); -// return NO_INIT; -// } -// } - - LOGD("Buffer size: %d", (int)bufferSize); - LOGD("Period size: %d", (int)periodSize); - LOGD("Latency: %d", (int)latency); - - mDefaults->bufferSize = bufferSize; - mDefaults->latency = latency; - mDefaults->periodSize = periodSize; - - // Commit the hardware parameters back to the device. - err = snd_pcm_hw_params(mHandle, mHardwareParams); - if (err < 0) { - LOGE("Unable to set hardware parameters: %s", snd_strerror(err)); - return NO_INIT; - } - - status = setSoftwareParams(); - - return status; -} - -const char *ALSAStreamOps::deviceName(int mode, uint32_t device) -{ - static char devString[ALSA_NAME_MAX]; - int dev; - int hasDevExt = 0; - - strcpy (devString, mDefaults->devicePrefix); - - for (dev=0; device; dev++) - if (device & (1 << dev)) { - /* Don't go past the end of our list */ - if (dev >= deviceSuffixLen) - break; - ALSA_STRCAT (devString, deviceSuffix[dev]); - device &= ~(1 << dev); - hasDevExt = 1; - } - - if (hasDevExt) - switch (mode) { - case AudioSystem::MODE_NORMAL: - ALSA_STRCAT (devString, "_normal"); - break; - case AudioSystem::MODE_RINGTONE: - ALSA_STRCAT (devString, "_ringtone"); - break; - case AudioSystem::MODE_IN_CALL: - ALSA_STRCAT (devString, "_incall"); - break; - }; - - return devString; -} - -// ---------------------------------------------------------------------------- - -AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) : - mParent(parent), - mPowerLock(false) -{ - static StreamDefaults _defaults = { - devicePrefix : "AndroidPlayback", - direction : SND_PCM_STREAM_PLAYBACK, - format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT - channelCount : 2, - sampleRate : DEFAULT_SAMPLE_RATE, - bufferRatio : 1, - latency : LATENCY_PLAYBACK_MS, // Desired Delay in usec - bufferSize : BUFFER_SZ_PLAYBACK, // Desired Number of samples - periodSize : PERIOD_SZ_PLAYBACK -}; - - setStreamDefaults(&_defaults); -} - -AudioStreamOutALSA::~AudioStreamOutALSA() -{ - standby(); -} - - -/* New arch */ -status_t AudioStreamOutALSA::setVolume(float left, float right) -{ - if (! mParent->mMixer || ! mDevice) - return NO_INIT; - - /** Tushar - Need to decide on the volume value - * that we pass onto the mixer. */ - return mParent->mMixer->setVolume (mDevice, (left + right)/2); -} - -status_t AudioStreamOutALSA::setVolume(float volume) -{ - if (! mParent->mMixer || ! mDevice) - return NO_INIT; - - return mParent->mMixer->setVolume (mDevice, volume); -} - -/* New Arch */ -status_t AudioStreamOutALSA::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - status_t status = NO_ERROR; - int device; - int value; - LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string()); - - if (param.getInt(String8(AudioParameter::keyRouting), device) == NO_ERROR) - { - mParent->doRouting(device); - - param.remove(String8(AudioParameter::keyRouting)); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -} - - -String8 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)mDevice); - } - - LOGD("AudioStreamOutALSA::getParameters() %s", param.toString().string()); - return param.toString(); -} - - -status_t AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames) -{ - - //TODO: enable when supported by driver - return INVALID_OPERATION; -} - - -ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes) -{ - snd_pcm_sframes_t n; - size_t sent = 0; - status_t err; - - mParent->lock().lock(); - AutoMutex lock(mLock); - - if (!mPowerLock) { - LOGD("Calling setDevice from write @..%d.\n",__LINE__); - ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK); - acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioOutLock"); - mPowerLock = true; - } - mParent->lock().unlock(); - - do { - // write correct number of bytes per attempt - n = snd_pcm_writei(mHandle, (char *) buffer + sent, snd_pcm_bytes_to_frames(mHandle, bytes - - sent)); - if (n == -EBADFD) { - LOGD("Calling setDevice.. pcm_write returned error @..%d.\n",__LINE__); - // Somehow the stream is in a bad state. The driver probably - // has a bug and snd_pcm_recover() doesn't seem to handle this. - ALSAStreamOps::setDevice(mParent->mode(), mDevice, PLAYBACK); - } else if (n < 0) { - if (mHandle) { - // snd_pcm_recover() will return 0 if successful in recovering from - // // an error, or -errno if the error was unrecoverable. - // We can make silent bit on as we are now handling the under-run and there will not be any data loss due to under-run - n = snd_pcm_recover(mHandle, n, 1); - if (n) - return static_cast<ssize_t> (n); - } - } else - sent += static_cast<ssize_t> (snd_pcm_frames_to_bytes(mHandle, n)); - } while (mHandle && sent < bytes); - //LOGI("Request Bytes=%d, Actual Written=%d",bytes,sent); - return snd_pcm_frames_to_bytes(mHandle, sent); -} - - -status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args) -{ - return NO_ERROR; -} - -status_t AudioStreamOutALSA::setDevice(int mode, - uint32_t newDevice, - uint32_t audio_mode, - bool force) -{ - AutoMutex lock(mLock); - - LOGV("AudioStreamOutALSA::setDevice(mode %d, newDevice %x, audio_mode %d), mDevice %x", - mode, newDevice, audio_mode, mDevice); - if (newDevice != mDevice || force) { - return ALSAStreamOps::setDevice(mode, newDevice, audio_mode); - } - return NO_ERROR; -} - -status_t AudioStreamOutALSA::standby() { - AutoMutex _l(mParent->lock()); - AutoMutex lock(mLock); - LOGD("Inside AudioStreamOutALSA::standby\n"); - - if (mParent->mode() != AudioSystem::MODE_IN_CALL) { - ALSAStreamOps::close(); - } - - if (mPowerLock) { - release_wake_lock("AudioOutLock"); - mPowerLock = false; - } - return NO_ERROR; -} - - -#define USEC_TO_MSEC(x) ((x + 999) / 1000) - -uint32_t AudioStreamOutALSA::latency() const -{ - // Android wants latency in milliseconds. - return USEC_TO_MSEC (mDefaults->latency); -} - -// ---------------------------------------------------------------------------- - -AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) : - mParent(parent), mPowerLock(false), - mDownSampler(NULL), mPcmIn(NULL) -{ - static StreamDefaults _defaults = { - devicePrefix : "AndroidRecord", - direction : SND_PCM_STREAM_CAPTURE, - format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT - channelCount : 1, - sampleRate : DEFAULT_SAMPLE_RATE, - bufferRatio : 1, - latency : LATENCY_CAPTURE_MS,// Desired Delay in usec - bufferSize : BUFFER_SZ_CAPTURE, // Desired Number of samples - periodSize : PERIOD_SZ_CAPTURE - }; - - setStreamDefaults(&_defaults); -} - -status_t AudioStreamInALSA::set(int *pformat, - uint32_t *pchannels, - uint32_t *prate) -{ - status_t status = ALSAStreamOps::set(pformat, pchannels, prate); - if (status == NO_ERROR && prate && *prate != DEFAULT_SAMPLE_RATE) { - mDownSampler = new ALSADownsampler(*prate, - mDefaults->channelCount, - PERIOD_SZ_CAPTURE, - this); - status = mDownSampler->initCheck(); - if (status != NO_ERROR) { - return status; - } - mPcmIn = new int16_t[PERIOD_SZ_CAPTURE * mDefaults->channelCount]; - } - return status; -} - -AudioStreamInALSA::~AudioStreamInALSA() -{ - standby(); - if (mDownSampler != NULL) { - delete mDownSampler; - } - if (mPcmIn != NULL) { - delete[] mPcmIn; - } -} - -status_t AudioStreamInALSA::setGain(float gain) -{ - if (mParent->mMixer) - return mParent->mMixer->setMasterGain (gain); - else - return NO_INIT; -} - -ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes) -{ - snd_pcm_sframes_t n; - - mParent->lock().lock(); - AutoMutex lock(mLock); - if (!mPowerLock) { - acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioInLock"); - - LOGD("Calling setDevice from read@..%d.\n",__LINE__); - ALSAStreamOps::setDevice(mParent->mode(), mDevice, CAPTURE); - - if (mDownSampler != NULL) { - mDownSampler->reset(); - mReadStatus = 0; - mInPcmInBuf = 0; - } - mPowerLock = true; - } - mParent->lock().unlock(); - - if (!mHandle) { - return -1; - } - - // FIXME: only support reads of exactly bufferSize() for now - if (bytes != (ssize_t)bufferSize()) { - LOGW("AudioStreamInALSA::read bad read size %d expected %d", (int)bytes, bufferSize()); - return -1; - } - - size_t frames = snd_pcm_bytes_to_frames(mHandle, bytes); - do { - if (mDownSampler) { - status_t status = mDownSampler->resample((int16_t *)buffer, &frames); - if (status != NO_ERROR) { - if (mReadStatus != 0) { - n = mReadStatus; - } else { - n = status; - } - frames = snd_pcm_bytes_to_frames(mHandle, bytes); - } else { - n = frames; - } - } else { - n = snd_pcm_readi(mHandle, - (uint8_t *)buffer, - frames); - } - if (n < 0) { - LOGD("AudioStreamInALSA::read error %d", (int)n); - n = snd_pcm_recover(mHandle, n, 0); - LOGD("AudioStreamInALSA::snd_pcm_recover error %d", (int)n); - if (n) - return static_cast<ssize_t> (n); - } - } while (n == 0); - - return snd_pcm_frames_to_bytes(mHandle, n); -} - -status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args) -{ - return NO_ERROR; -} - -status_t AudioStreamInALSA::setDevice(int mode, - uint32_t newDevice, - uint32_t audio_mode, - bool force) -{ - AutoMutex lock(mLock); - - return ALSAStreamOps::setDevice(mode, newDevice, audio_mode); -} - -status_t AudioStreamInALSA::standby() -{ - AutoMutex _l(mParent->lock()); - AutoMutex lock(mLock); - - LOGD("Entering AudioStreamInALSA::standby\n"); - - ALSAStreamOps::close(); - - if (mPowerLock) { - release_wake_lock ("AudioInLock"); - mPowerLock = false; - } - - return NO_ERROR; -} - -/* New Arch */ -status_t AudioStreamInALSA::setParameters(const String8& keyValuePairs) -{ - AudioParameter param = AudioParameter(keyValuePairs); - String8 key = String8("vr_mode"); - status_t status = NO_ERROR; - int value; - LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); - - - if (param.getInt(key, value) == NO_ERROR) { - mParent->setVoiceRecordGain((value != 0)); - param.remove(key); - } - - key = String8(AudioParameter::keyRouting); - if (param.getInt(key, value) == NO_ERROR) { - if(mHandle != NULL && value != 0) - setDevice(mParent->mode(), value, CAPTURE); - param.remove(key); - } - - if (param.size()) { - status = BAD_VALUE; - } - return status; -} - - -String8 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)mDevice); - } - - LOGD("AudioStreamInALSA::getParameters() %s", param.toString().string()); - return param.toString(); -} - -status_t AudioStreamInALSA::getNextBuffer(ALSABufferProvider::Buffer* buffer) -{ - if (mHandle == NULL) { - buffer->raw = NULL; - buffer->frameCount = 0; - return NO_INIT; - } - - if (mInPcmInBuf == 0) { - while (mInPcmInBuf < PERIOD_SZ_CAPTURE) { - mReadStatus = snd_pcm_readi(mHandle, - (uint8_t *)mPcmIn + - (mInPcmInBuf * mDefaults->channelCount * sizeof(int16_t)), - PERIOD_SZ_CAPTURE - mInPcmInBuf); - if (mReadStatus < 0) { - buffer->raw = NULL; - buffer->frameCount = 0; - LOGV("resampler read error %d", mReadStatus); - return mReadStatus; - } - mInPcmInBuf += mReadStatus; - } - } - - buffer->frameCount = (buffer->frameCount > mInPcmInBuf) ? mInPcmInBuf : buffer->frameCount; - buffer->i16 = mPcmIn + (PERIOD_SZ_CAPTURE - mInPcmInBuf) * mDefaults->channelCount; - - return NO_ERROR; -} - -void AudioStreamInALSA::releaseBuffer(ALSABufferProvider::Buffer* buffer) -{ - mInPcmInBuf -= buffer->frameCount; -} - - -// ---------------------------------------------------------------------------- - -struct mixer_info_t -{ - mixer_info_t() : - elem(0), - min(SND_MIXER_VOL_RANGE_MIN), - max(SND_MIXER_VOL_RANGE_MAX), - mute(false) - { - } - - snd_mixer_elem_t *elem; - long min; - long max; - long volume; - bool mute; - char name[ALSA_NAME_MAX]; -}; - -static int initMixer (snd_mixer_t **mixer, const char *name) -{ - int err; - - if ((err = snd_mixer_open(mixer, 0)) < 0) { - LOGE("Unable to open mixer: %s", snd_strerror(err)); - return err; - } - - if ((err = snd_mixer_attach(*mixer, name)) < 0) { - LOGE("Unable to attach mixer to device %s: %s", - name, snd_strerror(err)); - - if ((err = snd_mixer_attach(*mixer, "hw:00")) < 0) { - LOGE("Unable to attach mixer to device default: %s", - snd_strerror(err)); - - snd_mixer_close (*mixer); - *mixer = NULL; - return err; - } - } - - if ((err = snd_mixer_selem_register(*mixer, NULL, NULL)) < 0) { - LOGE("Unable to register mixer elements: %s", snd_strerror(err)); - snd_mixer_close (*mixer); - *mixer = NULL; - return err; - } - - // Get the mixer controls from the kernel - if ((err = snd_mixer_load(*mixer)) < 0) { - LOGE("Unable to load mixer elements: %s", snd_strerror(err)); - snd_mixer_close (*mixer); - *mixer = NULL; - return err; - } - - return 0; -} - -typedef int (*hasVolume_t)(snd_mixer_elem_t*); - -static const hasVolume_t hasVolume[] = { - snd_mixer_selem_has_playback_volume, - snd_mixer_selem_has_capture_volume -}; - -typedef int (*getVolumeRange_t)(snd_mixer_elem_t*, long int*, long int*); - -static const getVolumeRange_t getVolumeRange[] = { - snd_mixer_selem_get_playback_volume_range, - snd_mixer_selem_get_capture_volume_range -}; - -typedef int (*setVolume_t)(snd_mixer_elem_t*, long int); - -static const setVolume_t setVol[] = { - snd_mixer_selem_set_playback_volume_all, - snd_mixer_selem_set_capture_volume_all -}; - -ALSAMixer::ALSAMixer() -{ - int err; - - initMixer (&mMixer[SND_PCM_STREAM_PLAYBACK], "AndroidPlayback"); - initMixer (&mMixer[SND_PCM_STREAM_CAPTURE], "AndroidRecord"); - - snd_mixer_selem_id_t *sid; - snd_mixer_selem_id_alloca(&sid); - - for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) { - - mixer_info_t *info = mixerMasterProp[i].mInfo = new mixer_info_t; - - property_get (mixerMasterProp[i].propName, - info->name, - mixerMasterProp[i].propDefault); - - for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]); - elem; - elem = snd_mixer_elem_next(elem)) { - - if (!snd_mixer_selem_is_active(elem)) - continue; - - snd_mixer_selem_get_id(elem, sid); - - // Find PCM playback volume control element. - const char *elementName = snd_mixer_selem_id_get_name(sid); - - if (hasVolume[i] (elem)) - LOGD ("Mixer: element name: '%s'", elementName); - - if (info->elem == NULL && - strcmp(elementName, info->name) == 0 && - hasVolume[i] (elem)) { - - info->elem = elem; - getVolumeRange[i] (elem, &info->min, &info->max); - info->volume = info->max; - setVol[i] (elem, info->volume); - if (i == SND_PCM_STREAM_PLAYBACK && - snd_mixer_selem_has_playback_switch (elem)) - snd_mixer_selem_set_playback_switch_all (elem, 1); - break; - } - } - - LOGD ("Mixer: master '%s' %s.", info->name, info->elem ? "found" : "not found"); - - for (int j = 0; mixerProp[j][i].routes; j++) { - - mixer_info_t *info = mixerProp[j][i].mInfo = new mixer_info_t; - - property_get (mixerProp[j][i].propName, - info->name, - mixerProp[j][i].propDefault); - - for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]); - elem; - elem = snd_mixer_elem_next(elem)) { - - if (!snd_mixer_selem_is_active(elem)) - continue; - - snd_mixer_selem_get_id(elem, sid); - - // Find PCM playback volume control element. - const char *elementName = snd_mixer_selem_id_get_name(sid); - - if (info->elem == NULL && - strcmp(elementName, info->name) == 0 && - hasVolume[i] (elem)) { - - info->elem = elem; - getVolumeRange[i] (elem, &info->min, &info->max); - info->volume = info->max; - setVol[i] (elem, info->volume); - if (i == SND_PCM_STREAM_PLAYBACK && - snd_mixer_selem_has_playback_switch (elem)) - snd_mixer_selem_set_playback_switch_all (elem, 1); - break; - } - } - LOGD ("Mixer: route '%s' %s.", info->name, info->elem ? "found" : "not found"); - } - } - LOGD("mixer initialized."); -} - -ALSAMixer::~ALSAMixer() -{ - for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) { - if (mMixer[i]) snd_mixer_close (mMixer[i]); - if (mixerMasterProp[i].mInfo) { - delete mixerMasterProp[i].mInfo; - mixerMasterProp[i].mInfo = NULL; - } - for (int j = 0; mixerProp[j][i].routes; j++) { - if (mixerProp[j][i].mInfo) { - delete mixerProp[j][i].mInfo; - mixerProp[j][i].mInfo = NULL; - } - } - } - LOGD("mixer destroyed."); -} - -status_t ALSAMixer::setMasterVolume(float volume) -{ - mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_PLAYBACK].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - long minVol = info->min; - long maxVol = info->max; - - // Make sure volume is between bounds. - long vol = minVol + volume * (maxVol - minVol); - if (vol > maxVol) vol = maxVol; - if (vol < minVol) vol = minVol; - - info->volume = vol; - snd_mixer_selem_set_playback_volume_all (info->elem, vol); - - return NO_ERROR; -} - -status_t ALSAMixer::setMasterGain(float gain) -{ - mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_CAPTURE].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - long minVol = info->min; - long maxVol = info->max; - - // Make sure volume is between bounds. - long vol = minVol + gain * (maxVol - minVol); - if (vol > maxVol) vol = maxVol; - if (vol < minVol) vol = minVol; - - info->volume = vol; - snd_mixer_selem_set_capture_volume_all (info->elem, vol); - - return NO_ERROR; -} - -status_t ALSAMixer::setVolume(uint32_t device, float volume) -{ - for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - long minVol = info->min; - long maxVol = info->max; - - // Make sure volume is between bounds. - long vol = minVol + volume * (maxVol - minVol); - if (vol > maxVol) vol = maxVol; - if (vol < minVol) vol = minVol; - - info->volume = vol; - snd_mixer_selem_set_playback_volume_all (info->elem, vol); - } - - return NO_ERROR; -} - -status_t ALSAMixer::setGain(uint32_t device, float gain) -{ - for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - long minVol = info->min; - long maxVol = info->max; - - // Make sure volume is between bounds. - long vol = minVol + gain * (maxVol - minVol); - if (vol > maxVol) vol = maxVol; - if (vol < minVol) vol = minVol; - - info->volume = vol; - snd_mixer_selem_set_capture_volume_all (info->elem, vol); - } - - return NO_ERROR; -} - -status_t ALSAMixer::setCaptureMuteState(uint32_t device, bool state) -{ - for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - if (snd_mixer_selem_has_capture_switch (info->elem)) { - - int err = snd_mixer_selem_set_capture_switch_all (info->elem, static_cast<int>(!state)); - if (err < 0) { - LOGE("Unable to %s capture mixer switch %s", - state ? "enable" : "disable", info->name); - return INVALID_OPERATION; - } - } - - info->mute = state; - } - - return NO_ERROR; -} - -status_t ALSAMixer::getCaptureMuteState(uint32_t device, bool *state) -{ - if (! state) return BAD_VALUE; - - for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - *state = info->mute; - return NO_ERROR; - } - - return BAD_VALUE; -} - -status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state) -{ - - LOGE("\n set playback mute device %d, state %d \n", device,state); - - for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - if (snd_mixer_selem_has_playback_switch (info->elem)) { - - int err = snd_mixer_selem_set_playback_switch_all (info->elem, static_cast<int>(!state)); - if (err < 0) { - LOGE("Unable to %s playback mixer switch %s", - state ? "enable" : "disable", info->name); - return INVALID_OPERATION; - } - } - - info->mute = state; - } - - return NO_ERROR; -} - -status_t ALSAMixer::getPlaybackMuteState(uint32_t device, bool *state) -{ - if (! state) return BAD_VALUE; - - for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++) - if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) { - - mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo; - if (!info || !info->elem) return INVALID_OPERATION; - - *state = info->mute; - return NO_ERROR; - } - - return BAD_VALUE; -} - -// ---------------------------------------------------------------------------- - -ALSAControl::ALSAControl(const char *device) -{ - snd_ctl_open(&mHandle, device, 0); -} - -ALSAControl::~ALSAControl() -{ - if (mHandle) snd_ctl_close(mHandle); -} - -status_t ALSAControl::get(const char *name, unsigned int &value, int index) -{ - if (!mHandle) return NO_INIT; - - snd_ctl_elem_id_t *id; - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *control; - - snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_info_alloca(&info); - snd_ctl_elem_value_alloca(&control); - - snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); - snd_ctl_elem_id_set_name(id, name); - snd_ctl_elem_info_set_id(info, id); - - int ret = snd_ctl_elem_info(mHandle, info); - if (ret < 0) return BAD_VALUE; - - snd_ctl_elem_info_get_id(info, id); - snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info); - unsigned int count = snd_ctl_elem_info_get_count(info); - if ((unsigned int)index >= count) return BAD_VALUE; - - snd_ctl_elem_value_set_id(control, id); - - ret = snd_ctl_elem_read(mHandle, control); - if (ret < 0) return BAD_VALUE; - - switch (type) { - case SND_CTL_ELEM_TYPE_BOOLEAN: - value = snd_ctl_elem_value_get_boolean(control, index); - break; - case SND_CTL_ELEM_TYPE_INTEGER: - value = snd_ctl_elem_value_get_integer(control, index); - break; - case SND_CTL_ELEM_TYPE_INTEGER64: - value = snd_ctl_elem_value_get_integer64(control, index); - break; - case SND_CTL_ELEM_TYPE_ENUMERATED: - value = snd_ctl_elem_value_get_enumerated(control, index); - break; - case SND_CTL_ELEM_TYPE_BYTES: - value = snd_ctl_elem_value_get_byte(control, index); - break; - default: - return BAD_VALUE; - } - - return NO_ERROR; -} - -status_t ALSAControl::set(const char *name, unsigned int value, int index) -{ - if (!mHandle) return NO_INIT; - - snd_ctl_elem_id_t *id; - snd_ctl_elem_info_t *info; - snd_ctl_elem_value_t *control; - - snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_info_alloca(&info); - snd_ctl_elem_value_alloca(&control); - - snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); - snd_ctl_elem_id_set_name(id, name); - snd_ctl_elem_info_set_id(info, id); - - int ret = snd_ctl_elem_info(mHandle, info); - if (ret < 0) return BAD_VALUE; - - snd_ctl_elem_info_get_id(info, id); - snd_ctl_elem_type_t type = snd_ctl_elem_info_get_type(info); - unsigned int count = snd_ctl_elem_info_get_count(info); - - if (index >= (int)count) return BAD_VALUE; - - if (index == -1) - index = 0; // Range over all of them - else - count = index + 1; // Just do the one specified - - snd_ctl_elem_value_set_id(control, id); - - for (unsigned int i = index; i < count; i++) - switch (type) { - case SND_CTL_ELEM_TYPE_BOOLEAN: - snd_ctl_elem_value_set_boolean(control, i, value); - break; - case SND_CTL_ELEM_TYPE_INTEGER: - snd_ctl_elem_value_set_integer(control, i, value); - break; - case SND_CTL_ELEM_TYPE_INTEGER64: - snd_ctl_elem_value_set_integer64(control, i, value); - break; - case SND_CTL_ELEM_TYPE_ENUMERATED: - snd_ctl_elem_value_set_enumerated(control, i, value); - break; - case SND_CTL_ELEM_TYPE_BYTES: - snd_ctl_elem_value_set_byte(control, i, value); - break; - default: - break; - } - ret = snd_ctl_elem_write(mHandle, control); - return (ret < 0) ? BAD_VALUE : NO_ERROR; -} - -//------------------------------------------------------------------------------ -// 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; - - uint32_t in_sample_num = 0; // 16.16 fixed point - const uint32_t step = (uint32_t)(step_float * 65536.0f + 0.5f); // 16.16 fixed point - for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) { - const uint32_t whole = in_sample_num >> 16; - const uint32_t frac = (in_sample_num & 0xffff); // 0.16 fixed point - const int32_t s1 = tmp[whole]; - const int32_t s2 = tmp[whole + 1]; - *output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 16)); - } - } - - 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; -} - - -ALSADownsampler::ALSADownsampler(uint32_t outSampleRate, - uint32_t channelCount, - uint32_t frameCount, - ALSABufferProvider* 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("ALSADownsampler() cstor SR %d channels %d frames %d", - mSampleRate, mChannelCount, mFrameCount); - - if (mSampleRate != 8000 && mSampleRate != 11025 && mSampleRate != 16000 && - mSampleRate != 22050) { - LOGW("ALSADownsampler 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; -} - -ALSADownsampler::~ALSADownsampler() -{ - 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 ALSADownsampler::reset() -{ - mInInBuf = 0; - mInTmpBuf = 0; - mInTmp2Buf = 0; - mOutBufPos = 0; - mInOutBuf = 0; -} - - -int ALSADownsampler::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"); - - ALSABufferProvider::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; -} - -}; // namespace android - |