diff options
Diffstat (limited to 'libaudio/AudioHardware.cpp')
| -rw-r--r-- | libaudio/AudioHardware.cpp | 2014 |
1 files changed, 2014 insertions, 0 deletions
diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp new file mode 100644 index 0000000..10409a1 --- /dev/null +++ b/libaudio/AudioHardware.cpp @@ -0,0 +1,2014 @@ +/* +** 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 + +// ---------------------------------------------------------------------------- + +AudioHardware::AudioHardware() : + mInit(false), + mMicMute(false), + mPcm(NULL), + mMixer(NULL), + mPcmOpenCnt(0), + mMixerOpenCnt(0), + mInCallAudioMode(false), + mInputSource("Default"), + mBluetoothNrec(true), + 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; + + // bump thread priority to speed up mutex acquisition + int priority = getpriority(PRIO_PROCESS, 0); + setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_URGENT_AUDIO); + + // Mutex acquisition order is always out -> in -> hw + AutoMutex lock(mLock); + + spOut = mOutput; + while (spOut != 0) { + if (!spOut->checkStandby()) { + int cnt = spOut->standbyCnt(); + 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->standbyCnt(); + 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 + + setpriority(PRIO_PROCESS, 0, priority); + + 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(String8("Default")); + 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"; + + 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"); + } + } + + 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) +{ + LOGD("### setVoiceVolume"); + + AutoMutex lock(mLock); + 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); + } + + return NO_ERROR; +} + +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 %s\n", mInputSource.string()); + 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: + return "HP_NO_MIC"; + case AudioSystem::DEVICE_OUT_WIRED_HEADSET: + 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(String8 source) +{ + LOGV("setInputSource_l(%s)", source.string()); + if (source != mInputSource) { + if ((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; + } + LOGV("mixer_ctl_select, Input Source, (%s)", source.string()); + TRACE_DRIVER_IN(DRV_MIXER_SEL) + mixer_ctl_select(ctl, source.string()); + 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) +{ +} + +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; + + { // 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->standbyCnt(); + 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) + { + 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; +} + +//------------------------------------------------------------------------------ +// 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) +{ +} + +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; + + { // 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->standbyCnt(); + 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; + String8 source; + + LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); + + if (mHardware == NULL) return NO_INIT; + + { + AutoMutex lock(mLock); + + if (param.get(String8(INPUT_SOURCE_KEY), source) == NO_ERROR) { + AutoMutex hwLock(mHardware->lock()); + + mHardware->openMixer_l(); + mHardware->setInputSource_l(source); + mHardware->closeMixer_l(); + + param.remove(String8(INPUT_SOURCE_KEY)); + } + + 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 ; +} + +//------------------------------------------------------------------------------ +// 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; +} + + +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 |