--- /dev/null
+/* 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.
+*/
+
+#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_TAG "AudioHardwareALSA"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware/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)
+{
+ return new android::AudioHardwareALSA();
+}
+
+} // extern "C"
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+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';
+ LOG(LOG_ERROR, "ALSALib", 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"}
+ }
+};
+
+// ----------------------------------------------------------------------------
+
+AudioHardwareALSA::AudioHardwareALSA() :
+ mOutput(0),
+ mInput(0)
+{
+ snd_lib_error_set_handler(&ALSAErrorHandler);
+ mMixer = new ALSAMixer;
+}
+
+AudioHardwareALSA::~AudioHardwareALSA()
+{
+ if (mOutput) delete mOutput;
+ if (mInput) delete mInput;
+ if (mMixer) delete mMixer;
+}
+
+status_t AudioHardwareALSA::initCheck()
+{
+ if (mMixer && mMixer->isValid())
+ return NO_ERROR;
+ else
+ return NO_INIT;
+}
+
+status_t AudioHardwareALSA::standby()
+{
+ if (mOutput)
+ return mOutput->standby();
+
+ 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;
+}
+
+status_t AudioHardwareALSA::getMicMute(bool *state)
+{
+ 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;
+}
+
+status_t AudioHardwareALSA::dump(int fd, const Vector<String16>& args)
+{
+ 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!");
+ }
+
+ 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 (!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;
+ }
+ }
+ }
+ }
+
+ 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
+{
+ int c;
+
+ c = ALSAStreamOps::channelCount();
+
+ // 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);
+ }
+
+ 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);
+
+ // 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);
+}
+
+const char *AudioStreamInALSA::deviceName(int mode, int device)
+{
+ static char devString[PROPERTY_VALUE_MAX];
+
+ strcpy (devString, mDefaults->deviceName);
+ strcat (devString, "_Microphone");
+
+ return devString;
+}
+
+// ----------------------------------------------------------------------------
+
+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)
+{
+ 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 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;
+ }
+ }
+
+ 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;
+}
+
+status_t ALSAMixer::setVolume(mixer_types mixer, float volume)
+{
+ 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;
+}
+
+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)
+{
+ 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;
+
+ return NO_ERROR;
+}
+
+status_t ALSAMixer::setPlaybackMuteState(mixer_types mixer, 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;
+ }
+ }
+
+ info->mute = state;
+ return NO_ERROR;
+}
+
+status_t ALSAMixer::getPlaybackMuteState(mixer_types mixer, bool *state)
+{
+ 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
--- /dev/null
+/* AudioHardwareALSA.h
+**
+** 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.
+*/
+
+#ifndef ANDROID_AUDIO_HARDWARE_ALSA_H
+#define ANDROID_AUDIO_HARDWARE_ALSA_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <alsa/asoundlib.h>
+
+#include <hardware/AudioHardwareInterface.h>
+
+namespace android {
+
+class AudioHardwareALSA;
+
+// ----------------------------------------------------------------------------
+
+class ALSAMixer
+{
+public:
+ //
+ // Keep this in sync with AudioSystem::audio_routes
+ //
+ enum mixer_types {
+ MIXER_EARPIECE = 0,
+ MIXER_SPEAKER = 1,
+ MIXER_BLUETOOTH = 2,
+ MIXER_HEADSET = 3,
+ MIXER_LAST = MIXER_HEADSET
+ };
+
+ ALSAMixer();
+ virtual ~ALSAMixer();
+
+ bool isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
+ status_t setMasterVolume(float volume);
+ status_t setMasterGain(float gain);
+
+ status_t setVolume(mixer_types mixer, float volume);
+ status_t setGain(mixer_types mixer, float gain);
+
+ status_t setCaptureMuteState(mixer_types mixer, bool state);
+ status_t getCaptureMuteState(mixer_types mixer, bool *state);
+ status_t setPlaybackMuteState(mixer_types mixer, bool state);
+ status_t getPlaybackMuteState(mixer_types mixer, bool *state);
+
+private:
+ snd_mixer_t *mMixer[SND_PCM_STREAM_LAST+1];
+
+ struct mixer_info_t;
+ mixer_info_t *mMaster[SND_PCM_STREAM_LAST+1];
+ mixer_info_t *mInfo[SND_PCM_STREAM_LAST+1][MIXER_LAST+1];
+};
+
+class ALSAStreamOps
+{
+public:
+ struct StreamDefaults
+ {
+ const char * deviceName;
+ snd_pcm_stream_t direction; // playback or capture
+ snd_pcm_format_t format;
+ int channels;
+ uint32_t sampleRate;
+ unsigned int bufferTime; // Ring buffer length in usec
+ unsigned int periodTime; // Period time in usec
+ };
+
+ ALSAStreamOps();
+ virtual ~ALSAStreamOps();
+
+ status_t set(int format,
+ int channels,
+ uint32_t rate);
+ virtual uint32_t sampleRate() const;
+ status_t sampleRate(uint32_t rate);
+ virtual size_t bufferSize() const;
+ virtual int format() const;
+ virtual int channelCount() const;
+ status_t channelCount(int channels);
+ const char *streamName();
+ virtual status_t setDevice(int mode, uint32_t device);
+
+ virtual const char *deviceName(int mode, int device) = 0;
+
+protected:
+ friend class AudioStreamOutALSA;
+ friend class AudioStreamInALSA;
+
+ status_t open(int mode, int device);
+ void close();
+ status_t setSoftwareParams();
+ status_t setPCMFormat(snd_pcm_format_t format);
+ status_t setHardwareResample(bool resample);
+
+ void setStreamDefaults(StreamDefaults *dev)
+ {
+ mDefaults = dev;
+ }
+
+ Mutex mLock;
+
+private:
+ snd_pcm_t *mHandle;
+ snd_pcm_hw_params_t *mHardwareParams;
+ snd_pcm_sw_params_t *mSoftwareParams;
+ int mMode;
+ int mDevice;
+
+ StreamDefaults *mDefaults;
+};
+
+// ----------------------------------------------------------------------------
+
+class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
+{
+public:
+ AudioStreamOutALSA(AudioHardwareALSA *parent);
+ virtual ~AudioStreamOutALSA();
+
+ status_t set(int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0)
+ {
+ return ALSAStreamOps::set(format, channelCount, sampleRate);
+ }
+
+ virtual uint32_t sampleRate() const
+ {
+ return ALSAStreamOps::sampleRate();
+ }
+
+ virtual size_t bufferSize() const
+ {
+ return ALSAStreamOps::bufferSize();
+ }
+
+ virtual int channelCount() const;
+
+ virtual int format() const
+ {
+ return ALSAStreamOps::format();
+ }
+
+ virtual ssize_t write(const void *buffer, size_t bytes);
+ virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual status_t setDevice(int mode, uint32_t newDevice);
+
+ status_t setVolume(float volume);
+
+ virtual const char *deviceName(int mode, int device);
+
+ status_t standby();
+ bool isStandby();
+
+private:
+ AudioHardwareALSA *mParent;
+ bool mPowerLock;
+};
+
+
+class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
+{
+public:
+ AudioStreamInALSA(AudioHardwareALSA *parent);
+ virtual ~AudioStreamInALSA();
+
+ status_t set(int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0)
+ {
+ return ALSAStreamOps::set(format, channelCount, sampleRate);
+ }
+
+ virtual uint32_t sampleRate()
+ {
+ return ALSAStreamOps::sampleRate();
+ }
+
+ virtual size_t bufferSize() const
+ {
+ return ALSAStreamOps::bufferSize();
+ }
+
+ virtual int channelCount() const
+ {
+ return ALSAStreamOps::channelCount();
+ }
+
+ virtual int format() const
+ {
+ return ALSAStreamOps::format();
+ }
+
+ virtual ssize_t read(void* buffer, ssize_t bytes);
+ virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual status_t setDevice(int mode, uint32_t newDevice);
+
+ virtual status_t setGain(float gain);
+
+ virtual const char *deviceName(int mode, int device);
+
+private:
+ AudioHardwareALSA *mParent;
+};
+
+
+class AudioHardwareALSA : public AudioHardwareInterface
+{
+public:
+ AudioHardwareALSA();
+ virtual ~AudioHardwareALSA();
+
+ virtual status_t initCheck();
+ virtual status_t standby();
+ virtual status_t setVoiceVolume(float volume);
+ virtual status_t setMasterVolume(float volume);
+
+ virtual AudioStreamOut *openOutputStream(int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0);
+
+ virtual AudioStreamIn *openInputStream (int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0);
+
+ // Microphone mute
+ virtual status_t setMicMute(bool state);
+ virtual status_t getMicMute(bool *state);
+
+protected:
+ // audio routing
+ virtual status_t doRouting();
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+ friend class AudioStreamOutALSA;
+ friend class AudioStreamInALSA;
+
+ ALSAMixer *mMixer;
+ AudioStreamOutALSA *mOutput;
+ AudioStreamInALSA *mInput;
+
+private:
+ Mutex mLock;
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_AUDIO_HARDWARE_ALSA_H
--- /dev/null
+
+ Copyright (c) 2005-2008, The Android Open Source Project
+ 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.
+
+ 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.
+
+
+ Apache License
+ Version 2.0, January 2004
+ http://www.apache.org/licenses/
+
+ TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
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+
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+
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