/* AudioHardwareALSA.cpp
-**
-** Copyright 2008 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.
-*/
+ **
+ ** Copyright 2008-2010 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>
+#include <dlfcn.h>
#define LOG_TAG "AudioHardwareALSA"
#include <utils/Log.h>
#include <cutils/properties.h>
#include <media/AudioRecord.h>
-#include <hardware/power.h>
+#include <hardware_legacy/power.h>
-#include <alsa/asoundlib.h>
#include "AudioHardwareALSA.h"
-#define SND_MIXER_VOL_RANGE_MIN (0)
-#define SND_MIXER_VOL_RANGE_MAX (1000)
-
-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)
+extern "C"
{
- return new android::AudioHardwareALSA();
-}
-
-} // extern "C"
-
+ //
+ // Function for dlsym() to look up for creating a new AudioHardwareInterface.
+ //
+ android::AudioHardwareInterface *createAudioHardware(void) {
+ return android::AudioHardwareALSA::create();
+ }
+} // extern "C"
-namespace android {
+namespace android
+{
// ----------------------------------------------------------------------------
-static const char _nullALSADeviceName[] = "NULL_Device";
-
static void ALSAErrorHandler(const char *file,
int line,
const char *function,
l = snprintf(buf, BUFSIZ, "%s:%i:(%s) ", file, line, function);
vsnprintf(buf + l, BUFSIZ - l, fmt, arg);
buf[BUFSIZ-1] = '\0';
- LOG(LOG_ERROR, "ALSALib", buf);
+ LOG(LOG_ERROR, "ALSALib", "%s", buf);
va_end(arg);
}
-// ----------------------------------------------------------------------------
-
-struct alsa_properties_t {
- const char *propName;
- const char *propDefault;
-};
-
-static const alsa_properties_t masterPlaybackProp = {
- "alsa.mixer.playback.master", "PCM"
-};
-
-static const alsa_properties_t masterCaptureProp = {
- "alsa.mixer.capture.master", "Capture"
-};
-
-/* The following table(s) need to match in order of the route bits
- */
-static const char *deviceSuffix[] = {
- /* ROUTE_EARPIECE */ "_Earpiece",
- /* ROUTE_SPEAKER */ "_Speaker",
- /* ROUTE_BLUETOOTH */ "_Bluetooth",
- /* ROUTE_HEADSET */ "_Headset",
-};
-
-static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *));
-
-static const alsa_properties_t
- mixerMasterProp[SND_PCM_STREAM_LAST+1] =
-{
- { "alsa.mixer.playback.master", "PCM" },
- { "alsa.mixer.capture.master", "Capture" }
-};
-
-static const alsa_properties_t
- mixerProp[SND_PCM_STREAM_LAST+1][ALSAMixer::MIXER_LAST+1] =
-{
- {
- {"alsa.mixer.playback.earpiece", "Earpiece"},
- {"alsa.mixer.playback.speaker", "Speaker"},
- {"alsa.mixer.playback.bluetooth", "Bluetooth"},
- {"alsa.mixer.playback.headset", "Headphone"}
- },
- {
- {"alsa.mixer.capture.earpiece", "Capture"},
- {"alsa.mixer.capture.speaker", ""},
- {"alsa.mixer.capture.bluetooth", "Bluetooth Capture"},
- {"alsa.mixer.capture.headset", "Capture"}
- }
-};
-
-// ----------------------------------------------------------------------------
+AudioHardwareInterface *AudioHardwareALSA::create() {
+ return new AudioHardwareALSA();
+}
AudioHardwareALSA::AudioHardwareALSA() :
- mOutput(0),
- mInput(0)
+ mALSADevice(0),
+ mAcousticDevice(0)
{
snd_lib_error_set_handler(&ALSAErrorHandler);
mMixer = new ALSAMixer;
+
+ hw_module_t *module;
+ int err = hw_get_module(ALSA_HARDWARE_MODULE_ID,
+ (hw_module_t const**)&module);
+
+ if (err == 0) {
+ hw_device_t* device;
+ err = module->methods->open(module, ALSA_HARDWARE_NAME, &device);
+ if (err == 0) {
+ mALSADevice = (alsa_device_t *)device;
+ mALSADevice->init(mALSADevice, mDeviceList);
+ } else
+ LOGE("ALSA Module could not be opened!!!");
+ } else
+ LOGE("ALSA Module not found!!!");
+
+ err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
+ (hw_module_t const**)&module);
+
+ if (err == 0) {
+ hw_device_t* device;
+ err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
+ if (err == 0)
+ mAcousticDevice = (acoustic_device_t *)device;
+ else
+ LOGE("Acoustics Module not found.");
+ }
}
AudioHardwareALSA::~AudioHardwareALSA()
{
- if (mOutput) delete mOutput;
- if (mInput) delete mInput;
if (mMixer) delete mMixer;
+ if (mALSADevice)
+ mALSADevice->common.close(&mALSADevice->common);
+ if (mAcousticDevice)
+ mAcousticDevice->common.close(&mAcousticDevice->common);
}
status_t AudioHardwareALSA::initCheck()
{
- if (mMixer && mMixer->isValid())
- return NO_ERROR;
- else
- return NO_INIT;
-}
+ if (!mALSADevice)
+ return NO_INIT;
-status_t AudioHardwareALSA::standby()
-{
- if (mOutput)
- return mOutput->standby();
+ if (!mMixer || !mMixer->isValid())
+ LOGW("ALSA Mixer is not valid. AudioFlinger will do software volume control.");
return NO_ERROR;
}
status_t AudioHardwareALSA::setVoiceVolume(float volume)
{
// The voice volume is used by the VOICE_CALL audio stream.
- if (mMixer)
- return mMixer->setVolume(ALSAMixer::MIXER_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(int format,
- int channelCount,
- uint32_t sampleRate)
-{
- AutoMutex lock(mLock);
-
- // only one output stream allowed
- if (mOutput)
- return 0;
-
- AudioStreamOutALSA *out = new AudioStreamOutALSA(this);
-
- if (out->set(format, channelCount, sampleRate) == NO_ERROR) {
- mOutput = out;
- // Some information is expected to be available immediately after
- // the device is open.
- uint32_t routes = mRoutes[mMode];
- mOutput->setDevice(mMode, routes);
- } else {
- delete out;
- }
-
- return mOutput;
-}
-
-AudioStreamIn *AudioHardwareALSA::openInputStream(int format,
- int channelCount,
- uint32_t sampleRate)
-{
- AutoMutex lock(mLock);
-
- // only one input stream allowed
- if (mInput)
- return 0;
-
- AudioStreamInALSA *in = new AudioStreamInALSA(this);
-
- if (in->set(format, channelCount, sampleRate) == NO_ERROR) {
- mInput = in;
- // Now, actually open the device. Only 1 route used
- mInput->setDevice(0, 0);
- } else {
- delete in;
- }
- return mInput;
-}
-
-status_t AudioHardwareALSA::doRouting()
-{
- uint32_t routes;
-
- AutoMutex lock(mLock);
-
- if (mOutput) {
- routes = mRoutes[mMode];
- return mOutput->setDevice(mMode, routes);
- }
- return NO_INIT;
-}
-
-status_t AudioHardwareALSA::setMicMute(bool state)
-{
- ALSAMixer::mixer_types mixer_type =
- static_cast<ALSAMixer::mixer_types>(ffs(AudioSystem::ROUTE_EARPIECE) - 1);
-
if (mMixer)
- return mMixer->setCaptureMuteState(mixer_type, state);
-
- return NO_INIT;
+ return mMixer->setVolume(AudioSystem::DEVICE_OUT_EARPIECE, volume, volume);
+ else
+ return INVALID_OPERATION;
}
-status_t AudioHardwareALSA::getMicMute(bool *state)
+status_t AudioHardwareALSA::setMasterVolume(float volume)
{
- ALSAMixer::mixer_types mixer_type =
- static_cast<ALSAMixer::mixer_types>(ffs(AudioSystem::ROUTE_EARPIECE) - 1);
-
if (mMixer)
- return mMixer->getCaptureMuteState(mixer_type, state);
-
- return NO_ERROR;
+ return mMixer->setMasterVolume(volume);
+ else
+ return INVALID_OPERATION;
}
-status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
+status_t AudioHardwareALSA::setMode(int mode)
{
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-ALSAStreamOps::ALSAStreamOps() :
- mHandle(0),
- mHardwareParams(0),
- mSoftwareParams(0),
- mMode(-1),
- mDevice(-1)
-{
- if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) {
- LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!");
- }
+ status_t status = NO_ERROR;
- 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 format,
- int channels,
- uint32_t rate)
-{
- if (channels != 0)
- mDefaults->channels = channels;
-
- if (rate != 0)
- mDefaults->sampleRate = rate;
-
- switch(format) {
- case AudioSystem::DEFAULT: // format == 0
- break;
-
- case AudioSystem::PCM_16_BIT:
- mDefaults->format = SND_PCM_FORMAT_S16_LE;
- break;
-
- case AudioSystem::PCM_8_BIT:
- mDefaults->format = SND_PCM_FORMAT_S8;
- break;
-
- default:
- LOGE("Unknown PCM format %i. Forcing default", format);
- break;
- }
-
- return NO_ERROR;
-}
-
-uint32_t ALSAStreamOps::sampleRate() const
-{
- unsigned int rate;
- int err;
-
- if (! mHandle)
- return NO_INIT;
-
- return snd_pcm_hw_params_get_rate(mHardwareParams, &rate, 0) < 0
- ? 0 : static_cast<uint32_t>(rate);
-}
-
-status_t ALSAStreamOps::sampleRate(uint32_t rate)
-{
- const char *stream;
- unsigned int requestedRate;
- int err;
+ if (mode != mMode) {
+ status = AudioHardwareBase::setMode(mode);
- 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
-{
- snd_pcm_uframes_t periodSize;
- int err;
-
- if (!mHandle)
- return -1;
-
- err = snd_pcm_hw_params_get_period_size(mHardwareParams,
- &periodSize,
- 0);
- if (err < 0)
- return -1;
-
- return static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle, periodSize));
-}
-
-int ALSAStreamOps::format() const
-{
- snd_pcm_format_t ALSAFormat;
- int pcmFormatBitWidth;
- int audioSystemFormat;
-
- if (!mHandle)
- return -1;
-
- 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;
-}
-
-int ALSAStreamOps::channelCount() const
-{
- unsigned int val;
- int err;
-
- if (!mHandle)
- return -1;
-
- err = snd_pcm_hw_params_get_channels(mHardwareParams, &val);
- if (err < 0) {
- LOGE("Unable to get device channel count: %s",
- snd_strerror(err));
- return -1;
- }
-
- return val;
-}
-
-status_t ALSAStreamOps::channelCount(int channels)
-{
- int err;
-
- if (!mHandle)
- return NO_INIT;
-
- err = snd_pcm_hw_params_set_channels(mHandle, mHardwareParams, channels);
- if (err < 0) {
- LOGE("Unable to set channel count to %i: %s",
- channels, snd_strerror(err));
- return BAD_VALUE;
- }
-
- LOGD("Using %i %s for %s.",
- channels, channels == 1 ? "channel" : "channels", streamName());
-
- return NO_ERROR;
-}
-
-status_t ALSAStreamOps::open(int mode, int device)
-{
- const char *stream = streamName();
- const char *devName = deviceName(mode, device);
-
- int err;
-
- // 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.
- if ((err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0)) < 0) {
-
- // Try without the mode.
- devName = deviceName(AudioSystem::MODE_INVALID, device);
-
- err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
- if (err < 0) {
-
- // Try without mode or device.
- devName = deviceName(AudioSystem::MODE_INVALID, -1);
-
- err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
- if (err < 0) {
-
- err = snd_pcm_open(&mHandle, "hw:00,0", mDefaults->direction, 0);
-
- if (err < 0) {
- LOGE("Unable to open fallback %s device: %s",
- stream, snd_strerror(err));
-
- // Last resort is the NULL device (i.e. the bit bucket).
- err = snd_pcm_null_open(&mHandle, _nullALSADeviceName,
- mDefaults->direction, 0);
- if (err < 0) {
- LOG_FATAL("Unable to open NULL ALSA device: %s",
- snd_strerror(err));
- }
- LOGD("Opened NULL %s device.", streamName());
- return err;
- }
- }
+ if (status == NO_ERROR) {
+ // take care of mode change.
+ for(ALSAHandleList::iterator it = mDeviceList.begin();
+ it != mDeviceList.end(); ++it)
+ if (it->curDev) {
+ status = mALSADevice->route(&(*it), it->curDev, mode);
+ if (status != NO_ERROR)
+ break;
+ }
}
}
- mMode = mode;
- mDevice = device;
-
- LOGI("Initialized ALSA %s device %s", stream, devName);
- return err;
-}
-
-void ALSAStreamOps::close()
-{
- snd_pcm_t *handle = mHandle;
- mHandle = NULL;
-
- if (handle) {
- snd_pcm_close(handle);
- mMode = -1;
- mDevice = -1;
- }
-}
-
-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);
-
- 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;
- } 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)
-{
- // 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();
-
- status_t status = open (mode, device);
- int err;
-
- 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;
- }
-
- // 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;
- }
-
- status = setPCMFormat(mDefaults->format);
-
- //
- // 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->channels);
- if (status != NO_ERROR)
- return status;
-
- // Don't check for failure; some devices do not support the default
- // 44100 Hz rate.
- sampleRate(mDefaults->sampleRate);
-
- // Disable hardware resampling.
- status = setHardwareResample(false);
- if (status != NO_ERROR)
- return status;
-
- unsigned int bufferTime;
- unsigned int periodTime;
-
- // Set the buffer time.
- bufferTime = mDefaults->bufferTime;
- err = snd_pcm_hw_params_set_buffer_time_near(mHandle,
- mHardwareParams,
- &bufferTime,
- 0);
- if (err < 0) {
- LOGE("Unable to set buffer time to %u usec: %s",
- bufferTime, snd_strerror(err));
- return NO_INIT;
- }
-
- // Set the period time (i.e. the number of frames)
- periodTime = mDefaults->periodTime;
- err = snd_pcm_hw_params_set_period_time_near(mHandle,
- mHardwareParams,
- &periodTime,
- 0);
- if (err < 0) {
- LOGE("Unable to set period time to %u usec: %s",
- periodTime, snd_strerror(err));
- return NO_INIT;
- }
-
- // 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;
}
-// ----------------------------------------------------------------------------
-
-AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) :
- mParent(parent),
- mPowerLock(false)
-{
- static StreamDefaults _defaults =
- {
- deviceName : "AndroidPlayback",
- direction : SND_PCM_STREAM_PLAYBACK,
- format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
- channels : 2,
- sampleRate : 44100,
- bufferTime : 500000, // Ring buffer length in usec, 1/2 second
- periodTime : 100000, // Period time in usec
- };
-
- setStreamDefaults(&_defaults);
-}
-
-AudioStreamOutALSA::~AudioStreamOutALSA()
-{
- standby();
- mParent->mOutput = NULL;
-}
-
-int AudioStreamOutALSA::channelCount() const
+AudioStreamOut *
+AudioHardwareALSA::openOutputStream(uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
+ status_t *status)
{
- int c;
+ LOGD("openOutputStream called for devices: 0x%08x", devices);
- c = ALSAStreamOps::channelCount();
+ status_t err = BAD_VALUE;
+ AudioStreamOutALSA *out = 0;
- // AudioMixer will seg fault if it doesn't have two channels.
- LOGW_IF(c != 2,
- "AudioMixer expects two channels, but only %i found!", c);
- return c;
-}
-
-status_t AudioStreamOutALSA::setVolume(float volume)
-{
- if (! mParent->mMixer || mDevice < 0)
- return NO_INIT;
-
- ALSAMixer::mixer_types mixer_type = static_cast<ALSAMixer::mixer_types>(mDevice);
-
- return mParent->mMixer->setVolume (mixer_type, volume);
-}
-
-ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
-{
- snd_pcm_sframes_t n;
- status_t err;
-
- AutoMutex lock(mLock);
-
- if (isStandby())
- return 0;
-
- if (!mPowerLock) {
- acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioLock");
- ALSAStreamOps::setDevice(mMode, mDevice);
- mPowerLock = true;
- }
-
- n = snd_pcm_writei(mHandle,
- buffer,
- snd_pcm_bytes_to_frames(mHandle, bytes));
- if (n < 0 && mHandle) {
- // snd_pcm_recover() will return 0 if successful in recovering from
- // an error, or -errno if the error was unrecoverable.
- n = snd_pcm_recover(mHandle, n, 0);
+ if (devices & (devices - 1)) {
+ if (status) *status = err;
+ LOGD("openOutputStream called with bad devices");
+ return out;
}
- return static_cast<ssize_t>(n);
-}
-
-status_t AudioStreamOutALSA::dump(int fd, const Vector<String16>& args)
-{
- return NO_ERROR;
-}
-
-status_t AudioStreamOutALSA::setDevice(int mode, uint32_t newDevice)
-{
- uint32_t dev;
-
- //
- // Output to only one device. The new device is the first selected bit
- // in newDevice (per IAudioFlinger::ROUTE_*).
- //
- // It's possible to not output to any device (i.e. newDevice is 0).
- //
- dev = newDevice ? (ffs(static_cast<int>(newDevice)) - 1) : -1;
-
- AutoMutex lock(mLock);
-
- return ALSAStreamOps::setDevice(mode, dev);
-}
-
-const char *AudioStreamOutALSA::deviceName(int mode, int device)
-{
- static char devString[PROPERTY_VALUE_MAX];
- int hasDevExt = 0;
-
- strcpy (devString, mDefaults->deviceName);
-
- if (device >= 0 && device < deviceSuffixLen) {
- strcat (devString, deviceSuffix[device]);
- hasDevExt = 1;
- }
-
- if (hasDevExt)
- switch (mode) {
- case AudioSystem::MODE_NORMAL:
- strcat (devString, "_normal");
- break;
- case AudioSystem::MODE_RINGTONE:
- strcat (devString, "_ringtone");
- break;
- case AudioSystem::MODE_IN_CALL:
- strcat (devString, "_incall");
- break;
- };
-
- return devString;
-}
-
-status_t AudioStreamOutALSA::standby()
-{
- AutoMutex lock(mLock);
-
- if (mHandle)
- snd_pcm_drain (mHandle);
-
- if (mPowerLock) {
- release_wake_lock ("AudioLock");
- mPowerLock = false;
- }
-
- return NO_ERROR;
-}
-
-bool AudioStreamOutALSA::isStandby()
-{
- return (!mHandle);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) :
- mParent(parent)
-{
- static StreamDefaults _defaults =
- {
- deviceName : "AndroidRecord",
- direction : SND_PCM_STREAM_CAPTURE,
- format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
- channels : 1,
- sampleRate : AudioRecord::DEFAULT_SAMPLE_RATE,
- bufferTime : 500000, // Ring buffer length in usec, 1/2 second
- periodTime : 100000, // Period time in usec
- };
-
- setStreamDefaults(&_defaults);
-}
-
-AudioStreamInALSA::~AudioStreamInALSA()
-{
- mParent->mInput = NULL;
-}
-
-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;
- status_t err;
-
- AutoMutex lock(mLock);
-
- n = snd_pcm_readi(mHandle,
- buffer,
- snd_pcm_bytes_to_frames(mHandle, bytes));
- if (n < 0 && mHandle) {
- n = snd_pcm_recover(mHandle, n, 0);
- }
-
- return static_cast<ssize_t>(n);
-}
-
-status_t AudioStreamInALSA::dump(int fd, const Vector<String16>& args)
-{
- return NO_ERROR;
-}
-
-status_t AudioStreamInALSA::setDevice(int mode, uint32_t newDevice)
-{
- AutoMutex lock(mLock);
+ // Find the appropriate alsa device
+ for(ALSAHandleList::iterator it = mDeviceList.begin();
+ it != mDeviceList.end(); ++it)
+ if (it->devices & devices) {
+ err = mALSADevice->open(&(*it), devices, mode());
+ if (err) break;
+ out = new AudioStreamOutALSA(this, &(*it));
+ err = out->set(format, channels, sampleRate);
+ break;
+ }
- // The AudioHardwareALSA API does not allow one to set the input routing.
- // Only one input device (the microphone) is currently supported.
- //
- return ALSAStreamOps::setDevice(mode, AudioRecord::MIC_INPUT);
+ if (status) *status = err;
+ return out;
}
-const char *AudioStreamInALSA::deviceName(int mode, int device)
+void
+AudioHardwareALSA::closeOutputStream(AudioStreamOut* out)
{
- static char devString[PROPERTY_VALUE_MAX];
-
- strcpy (devString, mDefaults->deviceName);
- strcat (devString, "_Microphone");
-
- return devString;
+ delete out;
}
-// ----------------------------------------------------------------------------
-
-struct ALSAMixer::mixer_info_t {
- mixer_info_t() :
- elem(0), min(0), max(100), mute(false)
- {
- }
- snd_mixer_elem_t *elem;
- long min;
- long max;
- long volume;
- bool mute;
- char name[PROPERTY_VALUE_MAX];
-};
-
-static int initMixer (snd_mixer_t **mixer, const char *name)
+AudioStreamIn *
+AudioHardwareALSA::openInputStream(uint32_t devices,
+ int *format,
+ uint32_t *channels,
+ uint32_t *sampleRate,
+ status_t *status,
+ AudioSystem::audio_in_acoustics acoustics)
{
- int err;
+ status_t err = BAD_VALUE;
+ AudioStreamInALSA *in = 0;
- if ((err = snd_mixer_open(mixer, 0)) < 0) {
- LOGE("Unable to open mixer: %s", snd_strerror(err));
- return err;
+ if (devices & (devices - 1)) {
+ if (status) *status = err;
+ return in;
}
- 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 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 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 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++) {
-
- mMaster[i] = new mixer_info_t;
-
- property_get (mixerMasterProp[i].propName,
- mMaster[i]->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 (mMaster[i]->elem == NULL &&
- strcmp(elementName, mMaster[i]->name) == 0 &&
- hasVolume[i] (elem)) {
-
- mMaster[i]->elem = elem;
- getVolumeRange[i] (elem, &mMaster[i]->min, &mMaster[i]->max);
- mMaster[i]->volume = mMaster[i]->max;
- setVol[i] (elem, mMaster[i]->volume);
- if (i == SND_PCM_STREAM_PLAYBACK &&
- snd_mixer_selem_has_playback_switch (elem))
- snd_mixer_selem_set_playback_switch_all (elem, 1);
- break;
- }
+ // Find the appropriate alsa device
+ for(ALSAHandleList::iterator it = mDeviceList.begin();
+ it != mDeviceList.end(); ++it)
+ if (it->devices & devices) {
+ err = mALSADevice->open(&(*it), devices, mode());
+ if (err) break;
+ in = new AudioStreamInALSA(this, &(*it), acoustics);
+ err = in->set(format, channels, sampleRate);
+ break;
}
- for (int j = 0; j <= MIXER_LAST; j++) {
-
- mInfo[i][j] = new mixer_info_t;
-
- property_get (mixerProp[i][j].propName,
- mInfo[i][j]->name,
- mixerProp[i][j].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 (mInfo[i][j]->elem == NULL &&
- strcmp(elementName, mInfo[i][j]->name) == 0 &&
- hasVolume[i] (elem)) {
-
- mInfo[i][j]->elem = elem;
- getVolumeRange[i] (elem, &mInfo[i][j]->min, &mInfo[i][j]->max);
- mInfo[i][j]->volume = mInfo[i][j]->max;
- setVol[i] (elem, mInfo[i][j]->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 initialized.");
-}
-
-ALSAMixer::~ALSAMixer()
-{
- for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
- if (mMixer[i]) snd_mixer_close (mMixer[i]);
- if (mMaster[i]) delete mMaster[i];
- for (int j = 0; j <= MIXER_LAST; j++) {
- if (mInfo[i][j]) delete mInfo[i][j];
- }
- }
- LOGD("mixer destroyed.");
-}
-
-status_t ALSAMixer::setMasterVolume(float volume)
-{
- mixer_info_t *info = mMaster[SND_PCM_STREAM_PLAYBACK];
- 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 = mMaster[SND_PCM_STREAM_CAPTURE];
- 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;
+ if (status) *status = err;
+ return in;
}
-status_t ALSAMixer::setVolume(mixer_types mixer, float volume)
+void
+AudioHardwareALSA::closeInputStream(AudioStreamIn* in)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_PLAYBACK];
- 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(mixer_types mixer, float gain)
-{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_CAPTURE];
- 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;
+ delete in;
}
-status_t ALSAMixer::setCaptureMuteState(mixer_types mixer, bool state)
-{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_CAPTURE];
- if (!info || !info->elem) return INVALID_OPERATION;
-
- if (info->mute == state) return NO_ERROR;
-
- 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(mixer_types mixer, bool *state)
+status_t AudioHardwareALSA::setMicMute(bool state)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_CAPTURE];
- if (!info || !info->elem) return INVALID_OPERATION;
-
- if (! state) return BAD_VALUE;
-
- *state = info->mute;
+ if (mMixer)
+ return mMixer->setCaptureMuteState(AudioSystem::DEVICE_OUT_EARPIECE, state);
- return NO_ERROR;
+ return NO_INIT;
}
-status_t ALSAMixer::setPlaybackMuteState(mixer_types mixer, bool state)
+status_t AudioHardwareALSA::getMicMute(bool *state)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_PLAYBACK];
- 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;
- }
- }
+ if (mMixer)
+ return mMixer->getCaptureMuteState(AudioSystem::DEVICE_OUT_EARPIECE, state);
- info->mute = state;
return NO_ERROR;
}
-status_t ALSAMixer::getPlaybackMuteState(mixer_types mixer, bool *state)
+status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
{
- mixer_info_t *info = mInfo[SND_PCM_STREAM_PLAYBACK][mixer];
- if (!info || !info->elem) return INVALID_OPERATION;
-
- if (! state) return BAD_VALUE;
-
- *state = info->mute;
-
return NO_ERROR;
}
-// ----------------------------------------------------------------------------
-
-}; // namespace android
+} // namespace android
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