/* 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 2008-2009 Wind River Systems
+ **
+ ** Licensed under the Apache License, Version 2.0 (the "License");
+ ** you may not use this file except in compliance with the License.
+ ** You may obtain a copy of the License at
+ **
+ ** http://www.apache.org/licenses/LICENSE-2.0
+ **
+ ** Unless required by applicable law or agreed to in writing, software
+ ** distributed under the License is distributed on an "AS IS" BASIS,
+ ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ ** See the License for the specific language governing permissions and
+ ** limitations under the License.
+ */
#include <errno.h>
#include <stdarg.h>
#include <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" {
+#define SND_MIXER_VOL_RANGE_MAX (100)
-extern int ffs(int i);
+#define ALSA_NAME_MAX 128
-//
-// 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);
+#define ALSA_STRCAT(x,y) \
+ if (strlen(x) + strlen(y) < ALSA_NAME_MAX) \
+ strcat(x, y);
-//
-// Function for dlsym() to look up for creating a new AudioHardwareInterface.
-//
-android::AudioHardwareInterface *createAudioHardware(void)
+extern "C"
{
- return new android::AudioHardwareALSA();
-}
-} // 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);
-namespace android {
+ //
+ // Function for dlsym() to look up for creating a new AudioHardwareInterface.
+ //
+ android::AudioHardwareInterface *createAudioHardware(void) {
+ return new android::AudioHardwareALSA();
+ }
+
+} // extern "C"
+
+namespace android
+{
+
+typedef AudioSystem::audio_routes audio_routes;
+
+#define ROUTE_ALL AudioSystem::ROUTE_ALL
+#define ROUTE_EARPIECE AudioSystem::ROUTE_EARPIECE
+#define ROUTE_SPEAKER AudioSystem::ROUTE_SPEAKER
+#define ROUTE_BLUETOOTH_SCO AudioSystem::ROUTE_BLUETOOTH_SCO
+#define ROUTE_HEADSET AudioSystem::ROUTE_HEADSET
+#define ROUTE_BLUETOOTH_A2DP AudioSystem::ROUTE_BLUETOOTH_A2DP
// ----------------------------------------------------------------------------
// ----------------------------------------------------------------------------
-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",
+ /* ROUTE_EARPIECE */ "_Earpiece",
+ /* ROUTE_SPEAKER */ "_Speaker",
+ /* ROUTE_BLUETOOTH_SCO */ "_Bluetooth",
+ /* ROUTE_HEADSET */ "_Headset",
+ /* ROUTE_BLUETOOTH_A2DP */ "_Bluetooth-A2DP",
};
static const int deviceSuffixLen = (sizeof(deviceSuffix) / sizeof(char *));
-static const alsa_properties_t
- mixerMasterProp[SND_PCM_STREAM_LAST+1] =
+struct mixer_info_t;
+
+struct alsa_properties_t
{
- { "alsa.mixer.playback.master", "PCM" },
- { "alsa.mixer.capture.master", "Capture" }
+ const audio_routes routes;
+ const char *propName;
+ const char *propDefault;
+ mixer_info_t *mInfo;
};
-static const alsa_properties_t
- mixerProp[SND_PCM_STREAM_LAST+1][ALSAMixer::MIXER_LAST+1] =
-{
+static alsa_properties_t masterPlaybackProp = {
+ ROUTE_ALL, "alsa.mixer.playback.master", "PCM", NULL
+};
+
+static alsa_properties_t masterCaptureProp = {
+ ROUTE_ALL, "alsa.mixer.capture.master", "Capture", NULL
+};
+
+static alsa_properties_t
+mixerMasterProp[SND_PCM_STREAM_LAST+1] = {
+ { ROUTE_ALL, "alsa.mixer.playback.master", "PCM", NULL},
+ { ROUTE_ALL, "alsa.mixer.capture.master", "Capture", NULL}
+};
+
+static alsa_properties_t
+mixerProp[][SND_PCM_STREAM_LAST+1] = {
+ {
+ {ROUTE_EARPIECE, "alsa.mixer.playback.earpiece", "Earpiece", NULL},
+ {ROUTE_EARPIECE, "alsa.mixer.capture.earpiece", "Capture", NULL}
+ },
+ {
+ {ROUTE_SPEAKER, "alsa.mixer.playback.speaker", "Speaker", NULL},
+ {ROUTE_SPEAKER, "alsa.mixer.capture.speaker", "", NULL}
+ },
+ {
+ {ROUTE_BLUETOOTH_SCO, "alsa.mixer.playback.bluetooth.sco", "Bluetooth", NULL},
+ {ROUTE_BLUETOOTH_SCO, "alsa.mixer.capture.bluetooth.sco", "Bluetooth Capture", NULL}
+ },
+ {
+ {ROUTE_HEADSET, "alsa.mixer.playback.headset", "Headphone", NULL},
+ {ROUTE_HEADSET, "alsa.mixer.capture.headset", "Capture", NULL}
+ },
{
- {"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"}
- }
+ {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.playback.bluetooth.a2dp", "Bluetooth A2DP", NULL},
+ {ROUTE_BLUETOOTH_A2DP, "alsa.mixer.capture.bluetooth.a2dp", "Bluetooth A2DP Capture", NULL}
+ },
+ {
+ {static_cast<audio_routes>(0), NULL, NULL, NULL},
+ {static_cast<audio_routes>(0), NULL, NULL, NULL}
+ }
};
// ----------------------------------------------------------------------------
status_t AudioHardwareALSA::initCheck()
{
- if (mMixer && mMixer->isValid())
- return NO_ERROR;
- else
- return NO_INIT;
+ if (mMixer && mMixer->isValid())
+ return NO_ERROR;
+ else
+ return NO_INIT;
}
status_t AudioHardwareALSA::standby()
{
- if (mOutput)
- return mOutput->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;
+ if (mMixer)
+ return mMixer->setVolume(ROUTE_EARPIECE, volume);
+ else
+ return INVALID_OPERATION;
}
status_t AudioHardwareALSA::setMasterVolume(float volume)
{
- if (mMixer)
- return mMixer->setMasterVolume(volume);
- else
- return INVALID_OPERATION;
+ if (mMixer)
+ return mMixer->setMasterVolume(volume);
+ else
+ return INVALID_OPERATION;
}
-AudioStreamOut *AudioHardwareALSA::openOutputStream(int format,
- int channelCount,
- uint32_t sampleRate)
+AudioStreamOut *
+AudioHardwareALSA::openOutputStream(int format,
+ int channelCount,
+ uint32_t sampleRate,
+ status_t *status)
{
AutoMutex lock(mLock);
// only one output stream allowed
- if (mOutput)
+ if (mOutput) {
+ *status = ALREADY_EXISTS;
return 0;
+ }
AudioStreamOutALSA *out = new AudioStreamOutALSA(this);
- if (out->set(format, channelCount, sampleRate) == NO_ERROR) {
+ *status = out->set(format, channelCount, sampleRate);
+
+ if (*status == NO_ERROR) {
mOutput = out;
// Some information is expected to be available immediately after
// the device is open.
- uint32_t routes = mRoutes[mMode];
+ uint32_t routes = mRoutes[mMode];
mOutput->setDevice(mMode, routes);
- } else {
+ }
+ else {
delete out;
}
return mOutput;
}
-AudioStreamIn *AudioHardwareALSA::openInputStream(int format,
- int channelCount,
- uint32_t sampleRate)
+AudioStreamIn *
+AudioHardwareALSA::openInputStream(int format,
+ int channelCount,
+ uint32_t sampleRate,
+ status_t *status)
{
AutoMutex lock(mLock);
// only one input stream allowed
- if (mInput)
+ if (mInput) {
+ *status = ALREADY_EXISTS;
return 0;
+ }
AudioStreamInALSA *in = new AudioStreamInALSA(this);
- if (in->set(format, channelCount, sampleRate) == NO_ERROR) {
+ *status = in->set(format, channelCount, sampleRate);
+ if (*status == NO_ERROR) {
mInput = in;
// Now, actually open the device. Only 1 route used
mInput->setDevice(0, 0);
- } else {
+ }
+ else {
delete in;
}
return mInput;
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 mMixer->setCaptureMuteState(ROUTE_EARPIECE, 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 mMixer->getCaptureMuteState(ROUTE_EARPIECE, state);
return NO_ERROR;
}
mHardwareParams(0),
mSoftwareParams(0),
mMode(-1),
- mDevice(-1)
+ mDevice(0)
{
if (snd_pcm_hw_params_malloc(&mHardwareParams) < 0) {
LOG_ALWAYS_FATAL("Failed to allocate ALSA hardware parameters!");
ALSAStreamOps::~ALSAStreamOps()
{
- AutoMutex lock(mLock);
+ AutoMutex lock(mLock);
close();
mDefaults->sampleRate = rate;
switch(format) {
- case AudioSystem::DEFAULT: // format == 0
- break;
+ // format == 0
+ case AudioSystem::DEFAULT:
+ break;
- case AudioSystem::PCM_16_BIT:
- mDefaults->format = SND_PCM_FORMAT_S16_LE;
- 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;
+ case AudioSystem::PCM_8_BIT:
+ mDefaults->format = SND_PCM_FORMAT_S8;
+ break;
- default:
- LOGE("Unknown PCM format %i. Forcing default", format);
- break;
+ default:
+ LOGE("Unknown PCM format %i. Forcing default", format);
+ break;
}
- return NO_ERROR;
+ return NO_ERROR;
}
uint32_t ALSAStreamOps::sampleRate() const
return NO_INIT;
return snd_pcm_hw_params_get_rate(mHardwareParams, &rate, 0) < 0
- ? 0 : static_cast<uint32_t>(rate);
+ ? 0 : static_cast<uint32_t>(rate);
}
status_t ALSAStreamOps::sampleRate(uint32_t rate)
if (err < 0) {
LOGE("Unable to set %s sample rate to %u: %s",
- stream, rate, snd_strerror(err));
+ stream, rate, snd_strerror(err));
return BAD_VALUE;
}
if (requestedRate != rate) {
// 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 {
+ rate, requestedRate);
+ }
+ else {
LOGD("Set %s sample rate to %u HZ", stream, requestedRate);
}
return NO_ERROR;
//
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);
+ snd_pcm_uframes_t bufferSize = 0;
+ snd_pcm_uframes_t periodSize = 0;
+
+ err = snd_pcm_get_params(mHandle, &bufferSize, &periodSize);
+
if (err < 0)
return -1;
- return static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle, periodSize));
+ return static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle, bufferSize));
}
int ALSAStreamOps::format() const
pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat);
audioSystemFormat = AudioSystem::DEFAULT;
- switch(pcmFormatBitWidth)
- {
- case 8:
- audioSystemFormat = AudioSystem::PCM_8_BIT;
- break;
+ switch(pcmFormatBitWidth) {
+ case 8:
+ audioSystemFormat = AudioSystem::PCM_8_BIT;
+ break;
- case 16:
- audioSystemFormat = AudioSystem::PCM_16_BIT;
- break;
+ case 16:
+ audioSystemFormat = AudioSystem::PCM_16_BIT;
+ break;
- default:
- LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
+ default:
+ LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
}
return audioSystemFormat;
err = snd_pcm_hw_params_get_channels(mHardwareParams, &val);
if (err < 0) {
LOGE("Unable to get device channel count: %s",
- snd_strerror(err));
+ snd_strerror(err));
return -1;
}
return val;
}
-status_t ALSAStreamOps::channelCount(int channels)
-{
+status_t ALSAStreamOps::channelCount(int channels) {
int err;
if (!mHandle)
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));
+ channels, snd_strerror(err));
return BAD_VALUE;
}
LOGD("Using %i %s for %s.",
- channels, channels == 1 ? "channel" : "channels", streamName());
+ channels, channels == 1 ? "channel" : "channels", streamName());
return NO_ERROR;
}
-status_t ALSAStreamOps::open(int mode, int device)
+status_t ALSAStreamOps::open(int mode, uint32_t 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) {
+ for(;;) {
+ // The PCM stream is opened in blocking mode, per ALSA defaults. The
+ // AudioFlinger seems to assume blocking mode too, so asynchronous mode
+ // should not be used.
+ err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
+ if (err == 0) break;
- // Try without the mode.
- devName = deviceName(AudioSystem::MODE_INVALID, device);
+ // See if there is a less specific name we can try.
+ // Note: We are changing the contents of a const char * here.
+ char *tail = strrchr(devName, '_');
+ if (! tail) break;
+ *tail = 0;
+ }
+ if (err < 0) {
+ // None of the Android defined audio devices exist. Open a generic one.
+ devName = "hw:00,0";
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;
- }
- }
+ // Last resort is the NULL device (i.e. the bit bucket).
+ devName = _nullALSADeviceName;
+ err = snd_pcm_open(&mHandle, devName, mDefaults->direction, 0);
}
}
- mMode = mode;
- mDevice = device;
+ mMode = mode;
+ mDevice = device;
LOGI("Initialized ALSA %s device %s", stream, devName);
return err;
void ALSAStreamOps::close()
{
- snd_pcm_t *handle = mHandle;
+ snd_pcm_t *handle = mHandle;
mHandle = NULL;
if (handle) {
snd_pcm_close(handle);
- mMode = -1;
- mDevice = -1;
+ mMode = -1;
+ mDevice = 0;
}
}
// For playback, configure ALSA to start the transfer when the
// buffer is almost full.
startThreshold = (bufferSize / periodSize) * periodSize;
- } else {
+ }
+ 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);
+ mSoftwareParams,
+ startThreshold);
if (err < 0) {
LOGE("Unable to set start threshold to %lu frames: %s",
- startThreshold, snd_strerror(err));
+ startThreshold, snd_strerror(err));
return NO_INIT;
}
bufferSize);
if (err < 0) {
LOGE("Unable to set stop threshold to %lu frames: %s",
- bufferSize, snd_strerror(err));
+ bufferSize, snd_strerror(err));
return NO_INIT;
}
periodSize);
if (err < 0) {
LOGE("Unable to configure available minimum to %lu: %s",
- periodSize, snd_strerror(err));
+ periodSize, snd_strerror(err));
return NO_INIT;
}
err = snd_pcm_sw_params(mHandle, mSoftwareParams);
if (err < 0) {
LOGE("Unable to configure software parameters: %s",
- snd_strerror(err));
+ snd_strerror(err));
return NO_INIT;
}
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));
+ formatName, formatDesc, snd_strerror(err));
return NO_INIT;
}
static_cast<int>(resample));
if (err < 0) {
LOGE("Unable to %s hardware resampling: %s",
- resample ? "enable" : "disable",
- snd_strerror(err));
+ resample ? "enable" : "disable",
+ snd_strerror(err));
return NO_INIT;
}
return NO_ERROR;
const char *stream = streamName();
status_t status = open (mode, device);
- int err;
+ int err;
if (status != NO_ERROR)
return status;
return NO_INIT;
}
+ status = setPCMFormat(mDefaults->format);
+
// Set the interleaved read and write format.
err = snd_pcm_hw_params_set_access(mHandle, mHardwareParams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
LOGE("Unable to configure PCM read/write format: %s",
- snd_strerror(err));
+ 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.
if (status != NO_ERROR)
return status;
- unsigned int bufferTime;
- unsigned int periodTime;
+ snd_pcm_uframes_t bufferSize = mDefaults->bufferSize;
+ unsigned int latency = mDefaults->latency;
- // Set the buffer time.
- bufferTime = mDefaults->bufferTime;
- err = snd_pcm_hw_params_set_buffer_time_near(mHandle,
- mHardwareParams,
- &bufferTime,
- 0);
+ // Make sure we have at least the size we originally wanted
+ err = snd_pcm_hw_params_set_buffer_size(mHandle, mHardwareParams, bufferSize);
if (err < 0) {
- LOGE("Unable to set buffer time to %u usec: %s",
- bufferTime, snd_strerror(err));
+ LOGE("Unable to set buffer size to %d: %s",
+ (int)bufferSize, 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);
+ // Setup buffers for latency
+ err = snd_pcm_hw_params_set_buffer_time_near (mHandle, mHardwareParams,
+ &latency, NULL);
if (err < 0) {
- LOGE("Unable to set period time to %u usec: %s",
- periodTime, snd_strerror(err));
- return NO_INIT;
+ /* That didn't work, set the period instead */
+ unsigned int periodTime = latency / 4;
+ err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
+ &periodTime, NULL);
+ if (err < 0) {
+ LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
+ snd_pcm_uframes_t periodSize;
+ err = snd_pcm_hw_params_get_period_size (mHardwareParams, &periodSize, NULL);
+ if (err < 0) {
+ LOGE("Unable to get the period size for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
+ bufferSize = periodSize * 4;
+ if (bufferSize < mDefaults->bufferSize)
+ bufferSize = mDefaults->bufferSize;
+ err = snd_pcm_hw_params_set_buffer_size_near (mHandle, mHardwareParams, &bufferSize);
+ if (err < 0) {
+ LOGE("Unable to set the buffer size for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
+ } else {
+ // OK, we got buffer time near what we expect. See what that did for bufferSize.
+ err = snd_pcm_hw_params_get_buffer_size (mHardwareParams, &bufferSize);
+ if (err < 0) {
+ LOGE("Unable to get the buffer size for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
+ // Does set_buffer_time_near change the passed value? It should.
+ err = snd_pcm_hw_params_get_buffer_time (mHardwareParams, &latency, NULL);
+ if (err < 0) {
+ LOGE("Unable to get the buffer time for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
+ unsigned int periodTime = latency / 4;
+ err = snd_pcm_hw_params_set_period_time_near (mHandle, mHardwareParams,
+ &periodTime, NULL);
+ if (err < 0) {
+ LOGE("Unable to set the period time for latency: %s", snd_strerror(err));
+ return NO_INIT;
+ }
}
+ LOGD("Buffer size: %d", (int)bufferSize);
+ LOGD("Latency: %d", (int)latency);
+
+ mDefaults->bufferSize = bufferSize;
+ mDefaults->latency = latency;
+
// Commit the hardware parameters back to the device.
err = snd_pcm_hw_params(mHandle, mHardwareParams);
if (err < 0) {
// ----------------------------------------------------------------------------
AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent) :
- mParent(parent),
- mPowerLock(false)
+ mParent(parent),
+ mPowerLock(false)
{
- static StreamDefaults _defaults =
- {
- deviceName : "AndroidPlayback",
+ static StreamDefaults _defaults = {
+ devicePrefix : "AndroidPlayback",
direction : SND_PCM_STREAM_PLAYBACK,
- format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
+ 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
- };
+ latency : 250000, // Desired Delay in usec
+ bufferSize : 16384, // Desired Number of samples
+ };
setStreamDefaults(&_defaults);
}
AudioStreamOutALSA::~AudioStreamOutALSA()
{
- standby();
- mParent->mOutput = NULL;
+ standby();
+ mParent->mOutput = NULL;
}
int AudioStreamOutALSA::channelCount() const
{
- int c;
-
- c = ALSAStreamOps::channelCount();
+ int 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);
+ "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);
+ if (! mParent->mMixer || ! mDevice)
+ return NO_INIT;
- return mParent->mMixer->setVolume (mixer_type, volume);
+ return mParent->mMixer->setVolume (mDevice, volume);
}
ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
AutoMutex lock(mLock);
if (isStandby())
- return 0;
+ return 0;
if (!mPowerLock) {
acquire_wake_lock (PARTIAL_WAKE_LOCK, "AudioLock");
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);
+ return ALSAStreamOps::setDevice(mode, newDevice);
}
-const char *AudioStreamOutALSA::deviceName(int mode, int device)
+const char *AudioStreamOutALSA::deviceName(int mode, uint32_t device)
{
- static char devString[PROPERTY_VALUE_MAX];
- int hasDevExt = 0;
-
- strcpy (devString, mDefaults->deviceName);
+ static char devString[ALSA_NAME_MAX];
+ int dev;
+ int hasDevExt = 0;
+
+ strcpy (devString, mDefaults->devicePrefix);
+
+ for (dev=0; device; dev++)
+ if (device & (1 << dev)) {
+ /* Don't go past the end of our list */
+ if (dev >= deviceSuffixLen)
+ break;
+ ALSA_STRCAT (devString, deviceSuffix[dev]);
+ device &= ~(1 << dev);
+ hasDevExt = 1;
+ }
- if (device >= 0 && device < deviceSuffixLen) {
- strcat (devString, deviceSuffix[device]);
- hasDevExt = 1;
- }
+ if (hasDevExt)
+ switch (mode) {
+ case AudioSystem::MODE_NORMAL:
+ ALSA_STRCAT (devString, "_normal");
+ break;
+ case AudioSystem::MODE_RINGTONE:
+ ALSA_STRCAT (devString, "_ringtone");
+ break;
+ case AudioSystem::MODE_IN_CALL:
+ ALSA_STRCAT (devString, "_incall");
+ break;
+ };
- 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;
+ return devString;
}
status_t AudioStreamOutALSA::standby()
return (!mHandle);
}
+#define USEC_TO_MSEC(x) ((x + 999) / 1000)
+
+uint32_t AudioStreamOutALSA::latency() const
+{
+ // Android wants latency in milliseconds.
+ return USEC_TO_MSEC (mDefaults->latency);
+}
+
// ----------------------------------------------------------------------------
AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent) :
- mParent(parent)
+ mParent(parent)
{
- static StreamDefaults _defaults =
- {
- deviceName : "AndroidRecord",
+ static StreamDefaults _defaults = {
+ devicePrefix : "AndroidRecord",
direction : SND_PCM_STREAM_CAPTURE,
- format : SND_PCM_FORMAT_S16_LE, // AudioSystem::PCM_16_BIT
+ 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
- };
+ latency : 250000, // Desired Delay in usec
+ bufferSize : 16384, // Desired Number of samples
+ };
setStreamDefaults(&_defaults);
}
AudioStreamInALSA::~AudioStreamInALSA()
{
- mParent->mInput = NULL;
+ mParent->mInput = NULL;
}
status_t AudioStreamInALSA::setGain(float gain)
{
- if (mParent->mMixer)
- return mParent->mMixer->setMasterGain (gain);
- else
- return NO_INIT;
+ if (mParent->mMixer)
+ return mParent->mMixer->setMasterGain (gain);
+ else
+ return NO_INIT;
}
ssize_t AudioStreamInALSA::read(void *buffer, ssize_t bytes)
{
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);
+ return ALSAStreamOps::setDevice(mode, newDevice);
}
-const char *AudioStreamInALSA::deviceName(int mode, int device)
+const char *AudioStreamInALSA::deviceName(int mode, uint32_t device)
{
- static char devString[PROPERTY_VALUE_MAX];
+ static char devString[ALSA_NAME_MAX];
+
+ strcpy (devString, mDefaults->devicePrefix);
- strcpy (devString, mDefaults->deviceName);
+ // The AudioHardwareALSA API does not allow one to set the input routing.
+ // Only one input device (the microphone) is currently supported.
+ //
strcat (devString, "_Microphone");
return devString;
}
+status_t AudioStreamInALSA::standby()
+{
+ AutoMutex lock(mLock);
+
+ return NO_ERROR;
+}
+
// ----------------------------------------------------------------------------
-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];
+struct mixer_info_t
+{
+ mixer_info_t() :
+ elem(0),
+ min(SND_MIXER_VOL_RANGE_MIN),
+ max(SND_MIXER_VOL_RANGE_MAX),
+ mute(false)
+ {
+ }
+
+ snd_mixer_elem_t *elem;
+ long min;
+ long max;
+ long volume;
+ bool mute;
+ char name[ALSA_NAME_MAX];
};
static int initMixer (snd_mixer_t **mixer, const char *name)
{
- int err;
+ int err;
if ((err = snd_mixer_open(mixer, 0)) < 0) {
LOGE("Unable to open mixer: %s", snd_strerror(err));
if ((err = snd_mixer_attach(*mixer, name)) < 0) {
LOGE("Unable to attach mixer to device %s: %s",
- name, snd_strerror(err));
+ 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));
+ 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;
- }
+ 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;
+ 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;
+ snd_mixer_close (*mixer);
+ *mixer = NULL;
+ return err;
}
- return 0;
+ 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
+static const hasVolume_t hasVolume[] = {
+ snd_mixer_selem_has_playback_volume,
+ snd_mixer_selem_has_capture_volume
};
typedef int (*getVolumeRange_t)(snd_mixer_elem_t*, long int*, long int*);
-static getVolumeRange_t getVolumeRange[] =
-{
- snd_mixer_selem_get_playback_volume_range,
- snd_mixer_selem_get_capture_volume_range
+static const getVolumeRange_t getVolumeRange[] = {
+ snd_mixer_selem_get_playback_volume_range,
+ snd_mixer_selem_get_capture_volume_range
};
typedef int (*setVolume_t)(snd_mixer_elem_t*, long int);
-static setVolume_t setVol[] =
-{
- snd_mixer_selem_set_playback_volume_all,
- snd_mixer_selem_set_capture_volume_all
+static const setVolume_t setVol[] = {
+ snd_mixer_selem_set_playback_volume_all,
+ snd_mixer_selem_set_capture_volume_all
};
ALSAMixer::ALSAMixer()
{
int err;
- initMixer (&mMixer[SND_PCM_STREAM_PLAYBACK], "AndroidPlayback");
- initMixer (&mMixer[SND_PCM_STREAM_CAPTURE], "AndroidRecord");
+ 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++) {
+ for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
- mMaster[i] = new mixer_info_t;
+ mixer_info_t *info = mixerMasterProp[i].mInfo = new mixer_info_t;
- property_get (mixerMasterProp[i].propName,
- mMaster[i]->name,
- mixerMasterProp[i].propDefault);
+ property_get (mixerMasterProp[i].propName,
+ info->name,
+ mixerMasterProp[i].propDefault);
- for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
- elem;
- elem = snd_mixer_elem_next(elem)) {
+ 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;
+ if (!snd_mixer_selem_is_active(elem))
+ continue;
- snd_mixer_selem_get_id(elem, sid);
+ snd_mixer_selem_get_id(elem, sid);
- // Find PCM playback volume control element.
- const char *elementName = snd_mixer_selem_id_get_name(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)) {
+ if (hasVolume[i] (elem))
+ LOGD ("Mixer: element name: '%s'", elementName);
- 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;
- }
+ if (info->elem == NULL &&
+ strcmp(elementName, info->name) == 0 &&
+ hasVolume[i] (elem)) {
+
+ info->elem = elem;
+ getVolumeRange[i] (elem, &info->min, &info->max);
+ info->volume = info->max;
+ setVol[i] (elem, info->volume);
+ if (i == SND_PCM_STREAM_PLAYBACK &&
+ snd_mixer_selem_has_playback_switch (elem))
+ snd_mixer_selem_set_playback_switch_all (elem, 1);
+ break;
+ }
}
- for (int j = 0; j <= MIXER_LAST; j++) {
+ LOGD ("Mixer: master '%s' %s.", info->name, info->elem ? "found" : "not found");
+
+ for (int j = 0; mixerProp[j][i].routes; j++) {
- mInfo[i][j] = new mixer_info_t;
+ mixer_info_t *info = mixerProp[j][i].mInfo = new mixer_info_t;
- property_get (mixerProp[i][j].propName,
- mInfo[i][j]->name,
- mixerProp[i][j].propDefault);
+ property_get (mixerProp[j][i].propName,
+ info->name,
+ mixerProp[j][i].propDefault);
- for (snd_mixer_elem_t *elem = snd_mixer_first_elem(mMixer[i]);
- elem;
- elem = snd_mixer_elem_next(elem)) {
+ 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;
+ if (!snd_mixer_selem_is_active(elem))
+ continue;
- snd_mixer_selem_get_id(elem, sid);
+ snd_mixer_selem_get_id(elem, sid);
- // Find PCM playback volume control element.
- const char *elementName = snd_mixer_selem_id_get_name(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)) {
+ if (info->elem == NULL &&
+ strcmp(elementName, info->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.");
+ info->elem = elem;
+ getVolumeRange[i] (elem, &info->min, &info->max);
+ info->volume = info->max;
+ setVol[i] (elem, info->volume);
+ if (i == SND_PCM_STREAM_PLAYBACK &&
+ snd_mixer_selem_has_playback_switch (elem))
+ snd_mixer_selem_set_playback_switch_all (elem, 1);
+ break;
+ }
+ }
+ LOGD ("Mixer: route '%s' %s.", info->name, info->elem ? "found" : "not found");
+ }
+ }
+ LOGD("mixer initialized.");
}
ALSAMixer::~ALSAMixer()
{
- for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
- if (mMixer[i]) snd_mixer_close (mMixer[i]);
- if (mMaster[i]) delete mMaster[i];
- for (int j = 0; j <= MIXER_LAST; j++) {
- if (mInfo[i][j]) delete mInfo[i][j];
- }
- }
+ for (int i = 0; i <= SND_PCM_STREAM_LAST; i++) {
+ if (mMixer[i]) snd_mixer_close (mMixer[i]);
+ if (mixerMasterProp[i].mInfo) {
+ delete mixerMasterProp[i].mInfo;
+ mixerMasterProp[i].mInfo = NULL;
+ }
+ for (int j = 0; mixerProp[j][i].routes; j++) {
+ if (mixerProp[j][i].mInfo) {
+ delete mixerProp[j][i].mInfo;
+ mixerProp[j][i].mInfo = NULL;
+ }
+ }
+ }
LOGD("mixer destroyed.");
}
status_t ALSAMixer::setMasterVolume(float volume)
{
- mixer_info_t *info = mMaster[SND_PCM_STREAM_PLAYBACK];
+ mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_PLAYBACK].mInfo;
if (!info || !info->elem) return INVALID_OPERATION;
long minVol = info->min;
status_t ALSAMixer::setMasterGain(float gain)
{
- mixer_info_t *info = mMaster[SND_PCM_STREAM_CAPTURE];
+ mixer_info_t *info = mixerMasterProp[SND_PCM_STREAM_CAPTURE].mInfo;
if (!info || !info->elem) return INVALID_OPERATION;
long minVol = info->min;
return NO_ERROR;
}
-status_t ALSAMixer::setVolume(mixer_types mixer, float volume)
+status_t ALSAMixer::setVolume(uint32_t device, float volume)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_PLAYBACK];
- if (!info || !info->elem) return INVALID_OPERATION;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
- long minVol = info->min;
- long maxVol = info->max;
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
- // Make sure volume is between bounds.
- long vol = minVol + volume * (maxVol - minVol);
- if (vol > maxVol) vol = maxVol;
- if (vol < minVol) vol = minVol;
+ long minVol = info->min;
+ long maxVol = info->max;
- info->volume = vol;
- snd_mixer_selem_set_playback_volume_all (info->elem, vol);
+ // 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)
+status_t ALSAMixer::setGain(uint32_t device, float gain)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_CAPTURE];
- if (!info || !info->elem) return INVALID_OPERATION;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
- long minVol = info->min;
- long maxVol = info->max;
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
- // Make sure volume is between bounds.
- long vol = minVol + gain * (maxVol - minVol);
- if (vol > maxVol) vol = maxVol;
- if (vol < minVol) vol = minVol;
+ long minVol = info->min;
+ long maxVol = info->max;
- info->volume = vol;
- snd_mixer_selem_set_capture_volume_all (info->elem, vol);
+ // 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)
+status_t ALSAMixer::setCaptureMuteState(uint32_t device, bool state)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_CAPTURE];
- if (!info || !info->elem) return INVALID_OPERATION;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
- if (info->mute == state) return NO_ERROR;
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
- if (snd_mixer_selem_has_capture_switch (info->elem)) {
+ 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;
- }
- }
+ 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;
+ }
- info->mute = state;
return NO_ERROR;
}
-status_t ALSAMixer::getCaptureMuteState(mixer_types mixer, bool *state)
+status_t ALSAMixer::getCaptureMuteState(uint32_t device, 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;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_CAPTURE].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_CAPTURE].routes & device) {
- return NO_ERROR;
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_CAPTURE].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
+
+ *state = info->mute;
+ return NO_ERROR;
+ }
+
+ return BAD_VALUE;
}
-status_t ALSAMixer::setPlaybackMuteState(mixer_types mixer, bool state)
+status_t ALSAMixer::setPlaybackMuteState(uint32_t device, bool state)
{
- mixer_info_t *info = mInfo[mixer][SND_PCM_STREAM_PLAYBACK];
- if (!info || !info->elem) return INVALID_OPERATION;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
- if (snd_mixer_selem_has_playback_switch (info->elem)) {
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
- 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 (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;
+ }
- info->mute = state;
return NO_ERROR;
}
-status_t ALSAMixer::getPlaybackMuteState(mixer_types mixer, bool *state)
+status_t ALSAMixer::getPlaybackMuteState(uint32_t device, 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;
+ for (int j = 0; mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes; j++)
+ if (mixerProp[j][SND_PCM_STREAM_PLAYBACK].routes & device) {
- return NO_ERROR;
+ mixer_info_t *info = mixerProp[j][SND_PCM_STREAM_PLAYBACK].mInfo;
+ if (!info || !info->elem) return INVALID_OPERATION;
+
+ *state = info->mute;
+ return NO_ERROR;
+ }
+
+ return BAD_VALUE;
}
// ----------------------------------------------------------------------------
-}; // namespace android
+}; // namespace android
/* 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.
-*/
+ **
+ ** 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 <sys/types.h>
#include <alsa/asoundlib.h>
-#include <hardware/AudioHardwareInterface.h>
+#include <hardware_legacy/AudioHardwareBase.h>
-namespace android {
-
-class AudioHardwareALSA;
-
-// ----------------------------------------------------------------------------
-
-class ALSAMixer
+namespace android
{
-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);
+ class AudioHardwareALSA;
- 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
+ class ALSAMixer
{
- 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;
+ public:
+ ALSAMixer();
+ virtual ~ALSAMixer();
-private:
- snd_pcm_t *mHandle;
- snd_pcm_hw_params_t *mHardwareParams;
- snd_pcm_sw_params_t *mSoftwareParams;
- int mMode;
- int mDevice;
+ bool isValid() { return !!mMixer[SND_PCM_STREAM_PLAYBACK]; }
+ status_t setMasterVolume(float volume);
+ status_t setMasterGain(float gain);
- StreamDefaults *mDefaults;
-};
+ status_t setVolume(uint32_t device, float volume);
+ status_t setGain(uint32_t device, float gain);
-// ----------------------------------------------------------------------------
-
-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);
- }
+ status_t setCaptureMuteState(uint32_t device, bool state);
+ status_t getCaptureMuteState(uint32_t device, bool *state);
+ status_t setPlaybackMuteState(uint32_t device, bool state);
+ status_t getPlaybackMuteState(uint32_t device, bool *state);
- virtual uint32_t sampleRate() const
- {
- return ALSAStreamOps::sampleRate();
- }
+ private:
+ snd_mixer_t *mMixer[SND_PCM_STREAM_LAST+1];
+ };
- virtual size_t bufferSize() const
+ class ALSAStreamOps
{
- return ALSAStreamOps::bufferSize();
- }
+ public:
+ struct StreamDefaults
+ {
+ const char * devicePrefix;
+ snd_pcm_stream_t direction; // playback or capture
+ snd_pcm_format_t format;
+ int channels;
+ uint32_t sampleRate;
+ unsigned int latency; // Delay in usec
+ unsigned int bufferSize; // Size of sample buffer
+ };
+
+ 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, uint32_t device) = 0;
+
+ protected:
+ friend class AudioStreamOutALSA;
+ friend class AudioStreamInALSA;
+
+ status_t open(int mode, uint32_t 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;
+ uint32_t mDevice;
+
+ StreamDefaults *mDefaults;
+ };
- virtual int channelCount() const;
+ // ----------------------------------------------------------------------------
- virtual int format() const
+ class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps
{
- return ALSAStreamOps::format();
- }
+ public:
+ AudioStreamOutALSA(AudioHardwareALSA *parent);
+ virtual ~AudioStreamOutALSA();
- 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 set(int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0) {
+ return ALSAStreamOps::set(format, channelCount, sampleRate);
+ }
- status_t setVolume(float volume);
+ virtual uint32_t sampleRate() const
+ {
+ return ALSAStreamOps::sampleRate();
+ }
- virtual const char *deviceName(int mode, int device);
+ virtual size_t bufferSize() const
+ {
+ return ALSAStreamOps::bufferSize();
+ }
- status_t standby();
- bool isStandby();
+ virtual int channelCount() const;
-private:
- AudioHardwareALSA *mParent;
- bool mPowerLock;
-};
+ virtual int format() const
+ {
+ return ALSAStreamOps::format();
+ }
+ virtual uint32_t latency() const;
-class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
-{
-public:
- AudioStreamInALSA(AudioHardwareALSA *parent);
- virtual ~AudioStreamInALSA();
+ 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 set(int format = 0,
- int channelCount = 0,
- uint32_t sampleRate = 0)
- {
- return ALSAStreamOps::set(format, channelCount, sampleRate);
- }
+ status_t setVolume(float volume);
- virtual uint32_t sampleRate()
- {
- return ALSAStreamOps::sampleRate();
- }
+ virtual const char *deviceName(int mode, uint32_t device);
- virtual size_t bufferSize() const
- {
- return ALSAStreamOps::bufferSize();
- }
+ status_t standby();
+ bool isStandby();
- virtual int channelCount() const
- {
- return ALSAStreamOps::channelCount();
- }
+ private:
+ AudioHardwareALSA *mParent;
+ bool mPowerLock;
+ };
- virtual int format() const
+ class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
{
- return ALSAStreamOps::format();
- }
+ public:
+ AudioStreamInALSA(AudioHardwareALSA *parent);
+ virtual ~AudioStreamInALSA();
- 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);
+ status_t set(int format = 0,
+ int channelCount = 0,
+ uint32_t sampleRate = 0) {
+ return ALSAStreamOps::set(format, channelCount, sampleRate);
+ }
- virtual status_t setGain(float gain);
+ virtual uint32_t sampleRate() {
+ return ALSAStreamOps::sampleRate();
+ }
- virtual const char *deviceName(int mode, int device);
+ virtual size_t bufferSize() const
+ {
+ return ALSAStreamOps::bufferSize();
+ }
-private:
- AudioHardwareALSA *mParent;
-};
+ virtual int channelCount() const
+ {
+ return ALSAStreamOps::channelCount();
+ }
+ virtual int format() const
+ {
+ return ALSAStreamOps::format();
+ }
-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);
+ 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);
- // Microphone mute
- virtual status_t setMicMute(bool state);
- virtual status_t getMicMute(bool *state);
+ virtual status_t setGain(float gain);
-protected:
- // audio routing
- virtual status_t doRouting();
- virtual status_t dump(int fd, const Vector<String16>& args);
+ virtual const char *deviceName(int mode, uint32_t device);
- friend class AudioStreamOutALSA;
- friend class AudioStreamInALSA;
+ virtual status_t standby();
- ALSAMixer *mMixer;
- AudioStreamOutALSA *mOutput;
- AudioStreamInALSA *mInput;
-
-private:
- Mutex mLock;
-};
+ private:
+ AudioHardwareALSA *mParent;
+ };
-// ----------------------------------------------------------------------------
+ class AudioHardwareALSA : public AudioHardwareBase
+ {
+ public:
+ AudioHardwareALSA();
+ virtual ~AudioHardwareALSA();
+
+ /**
+ * check to see if the audio hardware interface has been initialized.
+ * return status based on values defined in include/utils/Errors.h
+ */
+ virtual status_t initCheck();
+
+ /**
+ * put the audio hardware into standby mode to conserve power. Returns
+ * status based on include/utils/Errors.h
+ */
+ virtual status_t standby();
+
+ /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+ virtual status_t setVoiceVolume(float volume);
+
+ /**
+ * set the audio volume for all audio activities other than voice call.
+ * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
+ * the software mixer will emulate this capability.
+ */
+ virtual status_t setMasterVolume(float volume);
+
+ // mic mute
+ virtual status_t setMicMute(bool state);
+ virtual status_t getMicMute(bool* state);
+
+ /** This method creates and opens the audio hardware output stream */
+ virtual AudioStreamOut* openOutputStream(
+ int format=0,
+ int channelCount=0,
+ uint32_t sampleRate=0,
+ status_t *status=0);
+
+ /** This method creates and opens the audio hardware input stream */
+ virtual AudioStreamIn* openInputStream(
+ int format,
+ int channelCount,
+ uint32_t sampleRate,
+ status_t *status);
+
+ protected:
+ /**
+ * doRouting actually initiates the routing. A call to setRouting
+ * or setMode may result in a routing change. The generic logic calls
+ * doRouting when required. If the device has any special requirements these
+ * methods can be overriden.
+ */
+ 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
+}; // namespace android
+#endif // ANDROID_AUDIO_HARDWARE_ALSA_H
OSDN Git Service
