diff options
Diffstat (limited to 'libaudio')
| -rw-r--r-- | libaudio/Android.mk | 10 | ||||
| -rw-r--r-- | libaudio/AudioHardware.cpp | 247 | ||||
| -rw-r--r-- | libaudio/AudioHardware.h | 7 | ||||
| -rw-r--r-- | libaudio/AudioPolicyManager.cpp | 1 | ||||
| -rw-r--r-- | libaudio/AudioPolicyManager.h | 8 | ||||
| -rw-r--r-- | libaudio/audio_policy.conf | 68 |
6 files changed, 197 insertions, 144 deletions
diff --git a/libaudio/Android.mk b/libaudio/Android.mk index d828e89..132ba9a 100644 --- a/libaudio/Android.mk +++ b/libaudio/Android.mk @@ -21,25 +21,21 @@ LOCAL_MODULE_TAGS := optional LOCAL_SHARED_LIBRARIES += libdl LOCAL_C_INCLUDES += \ external/tinyalsa/include \ - system/media/audio_effects/include \ - system/media/audio_utils/include + $(call include-path-for, audio-effects) \ + $(call include-path-for, audio-utils) include $(BUILD_SHARED_LIBRARY) include $(CLEAR_VARS) LOCAL_SRC_FILES := AudioPolicyManager.cpp -LOCAL_SHARED_LIBRARIES := libcutils libutils libmedia +LOCAL_SHARED_LIBRARIES := libcutils libutils LOCAL_STATIC_LIBRARIES := libmedia_helper LOCAL_WHOLE_STATIC_LIBRARIES := libaudiopolicy_legacy LOCAL_MODULE := audio_policy.aries LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw LOCAL_MODULE_TAGS := optional -ifeq ($(BOARD_HAVE_BLUETOOTH),true) - LOCAL_CFLAGS += -DWITH_A2DP -endif - ifeq ($(BOARD_HAVE_FM_RADIO),true) LOCAL_CFLAGS += -DHAVE_FM_RADIO endif diff --git a/libaudio/AudioHardware.cpp b/libaudio/AudioHardware.cpp index 4064a95..b14296e 100644 --- a/libaudio/AudioHardware.cpp +++ b/libaudio/AudioHardware.cpp @@ -32,7 +32,6 @@ #include <fcntl.h> #include "AudioHardware.h" -#include <media/AudioRecord.h> #include <audio_effects/effect_aec.h> #include <hardware_legacy/power.h> @@ -135,10 +134,10 @@ AudioHardware::~AudioHardware() if (mSecRilLibHandle) { if (disconnectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) - LOGE("Disconnect_RILD() error"); + ALOGE("Disconnect_RILD() error"); if (closeClientRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) - LOGE("CloseClient_RILD() error"); + ALOGE("CloseClient_RILD() error"); mRilClient = 0; @@ -159,7 +158,7 @@ void AudioHardware::loadRILD(void) mSecRilLibHandle = dlopen("libsecril-client.so", RTLD_NOW); if (mSecRilLibHandle) { - LOGV("libsecril-client.so is loaded"); + ALOGV("libsecril-client.so is loaded"); openClientRILD = (HRilClient (*)(void)) dlsym(mSecRilLibHandle, "OpenClient_RILD"); @@ -181,28 +180,28 @@ void AudioHardware::loadRILD(void) if (!openClientRILD || !disconnectRILD || !closeClientRILD || !isConnectedRILD || !connectRILD || !setCallVolume || !setCallAudioPath || !setCallClockSync) { - LOGE("Can't load all functions from libsecril-client.so"); + ALOGE("Can't load all functions from libsecril-client.so"); dlclose(mSecRilLibHandle); mSecRilLibHandle = NULL; } else { mRilClient = openClientRILD(); if (!mRilClient) { - LOGE("OpenClient_RILD() error"); + ALOGE("OpenClient_RILD() error"); dlclose(mSecRilLibHandle); mSecRilLibHandle = NULL; } } } else { - LOGE("Can't load libsecril-client.so"); + ALOGE("Can't load libsecril-client.so"); } } status_t AudioHardware::connectRILDIfRequired(void) { if (!mSecRilLibHandle) { - LOGE("connectIfRequired() lib is not loaded"); + ALOGE("connectIfRequired() lib is not loaded"); return INVALID_OPERATION; } @@ -211,7 +210,7 @@ status_t AudioHardware::connectRILDIfRequired(void) } if (connectRILD(mRilClient) != RIL_CLIENT_ERR_SUCCESS) { - LOGE("Connect_RILD() error"); + ALOGE("Connect_RILD() error"); return INVALID_OPERATION; } @@ -262,7 +261,7 @@ void AudioHardware::closeOutputStream(AudioStreamOut* out) { { Mutex::Autolock lock(mLock); if (mOutput == 0 || mOutput.get() != out) { - LOGW("Attempt to close invalid output stream"); + ALOGW("Attempt to close invalid output stream"); return; } spOut = mOutput; @@ -315,7 +314,7 @@ AudioStreamIn* AudioHardware::openInputStream( *status = rc; } - LOGV("AudioHardware::openInputStream()%p", in.get()); + ALOGV("AudioHardware::openInputStream()%p", in.get()); return in.get(); } @@ -327,13 +326,13 @@ void AudioHardware::closeInputStream(AudioStreamIn* in) { ssize_t index = mInputs.indexOf((AudioStreamInALSA *)in); if (index < 0) { - LOGW("Attempt to close invalid input stream"); + ALOGW("Attempt to close invalid input stream"); return; } spIn = mInputs[index]; mInputs.removeAt(index); } - LOGV("AudioHardware::closeInputStream()%p", in); + ALOGV("AudioHardware::closeInputStream()%p", in); spIn.clear(); } @@ -385,7 +384,7 @@ status_t AudioHardware::setMode(int mode) int prevMode = mMode; status = AudioHardwareBase::setMode(mode); - LOGV("setMode() : new %d, old %d", mMode, prevMode); + ALOGV("setMode() : new %d, old %d", mMode, prevMode); if (status == NO_ERROR) { // activate call clock in radio when entering in call mode if (mMode == AudioSystem::MODE_IN_CALL) @@ -398,15 +397,15 @@ status_t AudioHardware::setMode(int mode) if (mMode == AudioSystem::MODE_IN_CALL && !mInCallAudioMode) { if (spOut != 0) { - LOGV("setMode() in call force output standby"); + ALOGV("setMode() in call force output standby"); spOut->doStandby_l(); } if (spIn != 0) { - LOGV("setMode() in call force input standby"); + ALOGV("setMode() in call force input standby"); spIn->doStandby_l(); } - LOGV("setMode() openPcmOut_l()"); + ALOGV("setMode() openPcmOut_l()"); openPcmOut_l(); openMixer_l(); setInputSource_l(AUDIO_SOURCE_DEFAULT); @@ -420,22 +419,22 @@ status_t AudioHardware::setMode(int mode) struct mixer_ctl *ctl= mixer_get_ctl_by_name(mMixer, "Playback Path"); TRACE_DRIVER_OUT if (ctl != NULL) { - LOGV("setMode() reset Playback Path to RCV"); + ALOGV("setMode() reset Playback Path to RCV"); TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_set_enum_by_string(ctl, "RCV"); TRACE_DRIVER_OUT } } - LOGV("setMode() closePcmOut_l()"); + ALOGV("setMode() closePcmOut_l()"); closeMixer_l(); closePcmOut_l(); if (spOut != 0) { - LOGV("setMode() off call force output standby"); + ALOGV("setMode() off call force output standby"); spOut->doStandby_l(); } if (spIn != 0) { - LOGV("setMode() off call force input standby"); + ALOGV("setMode() off call force input standby"); spIn->doStandby_l(); } @@ -468,7 +467,7 @@ status_t AudioHardware::setMode(int mode) status_t AudioHardware::setMicMute(bool state) { - LOGV("setMicMute(%d) mMicMute %d", state, mMicMute); + ALOGV("setMicMute(%d) mMicMute %d", state, mMicMute); sp<AudioStreamInALSA> spIn; { AutoMutex lock(mLock); @@ -519,7 +518,7 @@ status_t AudioHardware::setParameters(const String8& keyValuePairs) mBluetoothNrec = true; } else { mBluetoothNrec = false; - LOGD("Turning noise reduction and echo cancellation off for BT " + ALOGD("Turning noise reduction and echo cancellation off for BT " "headset"); } param.remove(String8(BT_NREC_KEY)); @@ -541,7 +540,7 @@ status_t AudioHardware::setParameters(const String8& keyValuePairs) } if (ttyMode != mTTYMode) { - LOGV("new tty mode %d", ttyMode); + ALOGV("new tty mode %d", ttyMode); mTTYMode = ttyMode; if (mOutput != 0 && mMode == AudioSystem::MODE_IN_CALL) { setIncallPath_l(mOutput->device()); @@ -574,7 +573,7 @@ String8 AudioHardware::getParameters(const String8& keys) AudioParameter request = AudioParameter(keys); AudioParameter reply = AudioParameter(); - LOGV("getParameters() %s", keys.string()); + ALOGV("getParameters() %s", keys.string()); return reply.toString(); } @@ -582,16 +581,16 @@ String8 AudioHardware::getParameters(const String8& keys) size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) { if (format != AudioSystem::PCM_16_BIT) { - LOGW("getInputBufferSize bad format: %d", format); + ALOGW("getInputBufferSize bad format: %d", format); return 0; } if (channelCount < 1 || channelCount > 2) { - LOGW("getInputBufferSize bad channel count: %d", channelCount); + ALOGW("getInputBufferSize bad channel count: %d", channelCount); return 0; } if (sampleRate != getInputSampleRate(sampleRate)) { - LOGW("getInputBufferSize bad sample rate: %d", sampleRate); + ALOGW("getInputBufferSize bad sample rate: %d", sampleRate); return 0; } @@ -609,7 +608,7 @@ status_t AudioHardware::setVoiceVolume(float volume) void AudioHardware::setVoiceVolume_l(float volume) { - LOGD("### setVoiceVolume_l"); + ALOGD("### setVoiceVolume_l"); mVoiceVol = volume; @@ -623,33 +622,33 @@ void AudioHardware::setVoiceVolume_l(float volume) int int_volume = (int)(volume * 5); SoundType type; - LOGD("### route(%d) call volume(%f)", device, volume); + ALOGD("### route(%d) call volume(%f)", device, volume); switch (device) { case AudioSystem::DEVICE_OUT_EARPIECE: - LOGD("### earpiece call volume"); + ALOGD("### earpiece call volume"); type = SOUND_TYPE_VOICE; break; case AudioSystem::DEVICE_OUT_SPEAKER: - LOGD("### speaker call volume"); + ALOGD("### speaker call volume"); type = SOUND_TYPE_SPEAKER; break; case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - LOGD("### bluetooth call volume"); + ALOGD("### bluetooth call volume"); type = SOUND_TYPE_BTVOICE; break; case AudioSystem::DEVICE_OUT_WIRED_HEADSET: case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: // Use receive path with 3 pole headset. - LOGD("### headset call volume"); + ALOGD("### headset call volume"); type = SOUND_TYPE_HEADSET; break; default: - LOGW("### Call volume setting error!!!0x%08x \n", device); + ALOGW("### Call volume setting error!!!0x%08x \n", device); type = SOUND_TYPE_VOICE; break; } @@ -660,7 +659,7 @@ void AudioHardware::setVoiceVolume_l(float volume) status_t AudioHardware::setMasterVolume(float volume) { - LOGV("Set master volume to %f.\n", volume); + ALOGV("Set master volume to %f.\n", volume); // We return an error code here to let the audioflinger do in-software // volume on top of the maximum volume that we set through the SND API. // return error - software mixer will handle it @@ -673,9 +672,9 @@ status_t AudioHardware::setFmVolume(float v) mFmVolume = v; if (mFmFd > 0) { __u8 fmVolume = (AudioSystem::logToLinear(v) + 5) / 7; - LOGD("%s %f %d", __func__, v, (int) fmVolume); + ALOGD("%s %f %d", __func__, v, (int) fmVolume); if (ioctl(mFmFd, Si4709_IOC_VOLUME_SET, &fmVolume) < 0) { - LOGE("set_volume_fm error."); + ALOGE("set_volume_fm error."); return -EIO; } } @@ -759,30 +758,30 @@ status_t AudioHardware::dump(int fd, const Vector<String16>& args) status_t AudioHardware::setIncallPath_l(uint32_t device) { - LOGV("setIncallPath_l: device %x", device); + ALOGV("setIncallPath_l: device %x", device); // Setup sound path for CP clocking if ((mSecRilLibHandle) && (connectRILDIfRequired() == OK)) { if (mMode == AudioSystem::MODE_IN_CALL) { - LOGD("### incall mode route (%d)", device); + ALOGD("### incall mode route (%d)", device); AudioPath path; switch(device){ case AudioSystem::DEVICE_OUT_EARPIECE: - LOGD("### incall mode earpiece route"); + ALOGD("### incall mode earpiece route"); path = SOUND_AUDIO_PATH_HANDSET; break; case AudioSystem::DEVICE_OUT_SPEAKER: - LOGD("### incall mode speaker route"); + ALOGD("### incall mode speaker route"); path = SOUND_AUDIO_PATH_SPEAKER; break; case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - LOGD("### incall mode bluetooth route %s NR", mBluetoothNrec ? "" : "NO"); + ALOGD("### incall mode bluetooth route %s NR", mBluetoothNrec ? "" : "NO"); if (mBluetoothNrec) { path = SOUND_AUDIO_PATH_BLUETOOTH; } else { @@ -791,15 +790,15 @@ status_t AudioHardware::setIncallPath_l(uint32_t device) break; case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE : - LOGD("### incall mode headphone route"); + ALOGD("### incall mode headphone route"); path = SOUND_AUDIO_PATH_HEADPHONE; break; case AudioSystem::DEVICE_OUT_WIRED_HEADSET : - LOGD("### incall mode headset route"); + ALOGD("### incall mode headset route"); path = SOUND_AUDIO_PATH_HEADSET; break; default: - LOGW("### incall mode Error!! route = [%d]", device); + ALOGW("### incall mode Error!! route = [%d]", device); path = SOUND_AUDIO_PATH_HANDSET; break; } @@ -810,9 +809,9 @@ status_t AudioHardware::setIncallPath_l(uint32_t device) TRACE_DRIVER_IN(DRV_MIXER_GET) struct mixer_ctl *ctl= mixer_get_ctl_by_name(mMixer, "Voice Call Path"); TRACE_DRIVER_OUT - LOGE_IF(ctl == NULL, "setIncallPath_l() could not get mixer ctl"); + ALOGE_IF(ctl == NULL, "setIncallPath_l() could not get mixer ctl"); if (ctl != NULL) { - LOGV("setIncallPath_l() Voice Call Path, (%x)", device); + ALOGV("setIncallPath_l() Voice Call Path, (%x)", device); TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_set_enum_by_string(ctl, getVoiceRouteFromDevice(device)); TRACE_DRIVER_OUT @@ -825,10 +824,10 @@ status_t AudioHardware::setIncallPath_l(uint32_t device) #ifdef HAVE_FM_RADIO void AudioHardware::enableFMRadio() { - LOGV("AudioHardware::enableFMRadio() Turning FM Radio ON"); + ALOGV("AudioHardware::enableFMRadio() Turning FM Radio ON"); if (mMode == AudioSystem::MODE_IN_CALL) { - LOGV("AudioHardware::enableFMRadio() Call is active. Delaying FM enable."); + ALOGV("AudioHardware::enableFMRadio() Call is active. Delaying FM enable."); mFmResumeAfterCall = true; } else { @@ -837,7 +836,7 @@ void AudioHardware::enableFMRadio() { setInputSource_l(AUDIO_SOURCE_DEFAULT); if (mMixer != NULL) { - LOGV("AudioHardware::enableFMRadio() FM Radio is ON, calling setFMRadioPath_l()"); + ALOGV("AudioHardware::enableFMRadio() FM Radio is ON, calling setFMRadioPath_l()"); setFMRadioPath_l(mOutput->device()); } @@ -850,7 +849,7 @@ void AudioHardware::enableFMRadio() { } void AudioHardware::disableFMRadio() { - LOGV("AudioHardware::disableFMRadio() Turning FM Radio OFF"); + ALOGV("AudioHardware::disableFMRadio() Turning FM Radio OFF"); if (mMixer != NULL) { // Disable FM radio flag to allow the codec to be turned off @@ -878,7 +877,7 @@ void AudioHardware::disableFMRadio() { status_t AudioHardware::setFMRadioPath_l(uint32_t device) { - LOGV("setFMRadioPath_l() device %x", device); + ALOGV("setFMRadioPath_l() device %x", device); AudioPath path; const char *fmpath; @@ -889,45 +888,45 @@ status_t AudioHardware::setFMRadioPath_l(uint32_t device) * now is an ORed value and we need to get back its original value. */ device -= AudioSystem::DEVICE_OUT_SPEAKER; - LOGD("setFMRadioPath_l() device removed speaker %x", device); + ALOGD("setFMRadioPath_l() device removed speaker %x", device); } switch(device){ case AudioSystem::DEVICE_OUT_SPEAKER: - LOGD("setFMRadioPath_l() fmradio speaker route"); + ALOGD("setFMRadioPath_l() fmradio speaker route"); fmpath = "FMR_SPK"; break; case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET: case AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - LOGD("setFMRadioPath_l() fmradio bluetooth route"); + ALOGD("setFMRadioPath_l() fmradio bluetooth route"); break; case AudioSystem::DEVICE_OUT_WIRED_HEADSET: case AudioSystem::DEVICE_OUT_WIRED_HEADPHONE: - LOGD("setFMRadioPath_l() fmradio headphone route"); + ALOGD("setFMRadioPath_l() fmradio headphone route"); fmpath = "FMR_HP"; break; default: - LOGE("setFMRadioPath_l() fmradio error, route = [%d]", device); + ALOGE("setFMRadioPath_l() fmradio error, route = [%d]", device); fmpath = "FMR_HP"; break; } if (mMixer != NULL) { - LOGV("setFMRadioPath_l() mixer is open"); + ALOGV("setFMRadioPath_l() mixer is open"); TRACE_DRIVER_IN(DRV_MIXER_GET) struct mixer_ctl *ctl= mixer_get_control(mMixer, "FM Radio Path", 0); TRACE_DRIVER_OUT if (ctl != NULL) { - LOGV("setFMRadioPath_l() FM Radio Path, (%s)", fmpath); + ALOGV("setFMRadioPath_l() FM Radio Path, (%s)", fmpath); TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_select(ctl, fmpath); TRACE_DRIVER_OUT } else { - LOGE("setFMRadioPath_l() could not get FM Radio Path mixer ctl"); + ALOGE("setFMRadioPath_l() could not get FM Radio Path mixer ctl"); } TRACE_DRIVER_IN(DRV_MIXER_GET) @@ -935,17 +934,17 @@ status_t AudioHardware::setFMRadioPath_l(uint32_t device) TRACE_DRIVER_OUT const char *route = getOutputRouteFromDevice(device); - LOGV("setFMRadioPath_l() Playpack Path, (%s)", route); + ALOGV("setFMRadioPath_l() Playpack Path, (%s)", route); if (ctl) { TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_select(ctl, route); TRACE_DRIVER_OUT } else { - LOGE("setFMRadioPath_l() could not get Playback Path mixer ctl"); + ALOGE("setFMRadioPath_l() could not get Playback Path mixer ctl"); } } else { - LOGE("setFMRadioPath_l() mixer is not open"); + ALOGE("setFMRadioPath_l() mixer is not open"); } return NO_ERROR; @@ -954,10 +953,10 @@ status_t AudioHardware::setFMRadioPath_l(uint32_t device) struct pcm *AudioHardware::openPcmOut_l() { - LOGD("openPcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); + ALOGD("openPcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); if (mPcmOpenCnt++ == 0) { if (mPcm != NULL) { - LOGE("openPcmOut_l() mPcmOpenCnt == 0 and mPcm == %p\n", mPcm); + ALOGE("openPcmOut_l() mPcmOpenCnt == 0 and mPcm == %p\n", mPcm); mPcmOpenCnt--; return NULL; } @@ -978,7 +977,7 @@ struct pcm *AudioHardware::openPcmOut_l() mPcm = pcm_open(0, 0, flags, &config); TRACE_DRIVER_OUT if (!pcm_is_ready(mPcm)) { - LOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_get_error(mPcm)); + ALOGE("openPcmOut_l() cannot open pcm_out driver: %s\n", pcm_get_error(mPcm)); TRACE_DRIVER_IN(DRV_PCM_CLOSE) pcm_close(mPcm); TRACE_DRIVER_OUT @@ -991,9 +990,9 @@ struct pcm *AudioHardware::openPcmOut_l() void AudioHardware::closePcmOut_l() { - LOGD("closePcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); + ALOGD("closePcmOut_l() mPcmOpenCnt: %d", mPcmOpenCnt); if (mPcmOpenCnt == 0) { - LOGE("closePcmOut_l() mPcmOpenCnt == 0"); + ALOGE("closePcmOut_l() mPcmOpenCnt == 0"); return; } @@ -1007,10 +1006,10 @@ void AudioHardware::closePcmOut_l() struct mixer *AudioHardware::openMixer_l() { - LOGV("openMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); + ALOGV("openMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); if (mMixerOpenCnt++ == 0) { if (mMixer != NULL) { - LOGE("openMixer_l() mMixerOpenCnt == 0 and mMixer == %p\n", mMixer); + ALOGE("openMixer_l() mMixerOpenCnt == 0 and mMixer == %p\n", mMixer); mMixerOpenCnt--; return NULL; } @@ -1018,7 +1017,7 @@ struct mixer *AudioHardware::openMixer_l() mMixer = mixer_open(0); TRACE_DRIVER_OUT if (mMixer == NULL) { - LOGE("openMixer_l() cannot open mixer"); + ALOGE("openMixer_l() cannot open mixer"); mMixerOpenCnt--; return NULL; } @@ -1028,9 +1027,9 @@ struct mixer *AudioHardware::openMixer_l() void AudioHardware::closeMixer_l() { - LOGV("closeMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); + ALOGV("closeMixer_l() mMixerOpenCnt: %d", mMixerOpenCnt); if (mMixerOpenCnt == 0) { - LOGE("closeMixer_l() mMixerOpenCnt == 0"); + ALOGE("closeMixer_l() mMixerOpenCnt == 0"); return; } @@ -1160,7 +1159,7 @@ sp <AudioHardware::AudioStreamInALSA> AudioHardware::getActiveInput_l() status_t AudioHardware::setInputSource_l(audio_source source) { - LOGV("setInputSource_l(%d)", source); + ALOGV("setInputSource_l(%d)", source); if (source != mInputSource) { if ((source == AUDIO_SOURCE_DEFAULT) || (mMode != AudioSystem::MODE_IN_CALL)) { if (mMixer) { @@ -1189,7 +1188,7 @@ status_t AudioHardware::setInputSource_l(audio_source source) default: return NO_INIT; } - LOGV("mixer_ctl_set_enum_by_string, Input Source, (%s)", sourceName); + ALOGV("mixer_ctl_set_enum_by_string, Input Source, (%s)", sourceName); TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_set_enum_by_string(ctl, sourceName); TRACE_DRIVER_OUT @@ -1205,7 +1204,7 @@ struct echo_reference_itfe *AudioHardware::getEchoReference(audio_format_t forma uint32_t channelCount, uint32_t samplingRate) { - LOGV("AudioHardware::getEchoReference %p", mEchoReference); + ALOGV("AudioHardware::getEchoReference %p", mEchoReference); releaseEchoReference(mEchoReference); if (mOutput != NULL) { uint32_t wrChannelCount = popcount(mOutput->channels()); @@ -1227,7 +1226,7 @@ struct echo_reference_itfe *AudioHardware::getEchoReference(audio_format_t forma void AudioHardware::releaseEchoReference(struct echo_reference_itfe *reference) { - LOGV("AudioHardware::releaseEchoReference %p", mEchoReference); + ALOGV("AudioHardware::releaseEchoReference %p", mEchoReference); if (mEchoReference != NULL && reference == mEchoReference) { if (mOutput != NULL) { mOutput->removeEchoReference(reference); @@ -1302,7 +1301,7 @@ int AudioHardware::AudioStreamOutALSA::getPlaybackDelay(size_t frames, buffer->time_stamp.tv_sec = 0; buffer->time_stamp.tv_nsec = 0; buffer->delay_ns = 0; - LOGV("getPlaybackDelay(): pcm_get_htimestamp error, setting playbackTimestamp to 0"); + ALOGV("getPlaybackDelay(): pcm_get_htimestamp error, setting playbackTimestamp to 0"); return rc; } @@ -1313,7 +1312,7 @@ int AudioHardware::AudioStreamOutALSA::getPlaybackDelay(size_t frames, // sample being written. long delayNs = (long)(((int64_t)(kernelFr + frames)* 1000000000) /AUDIO_HW_OUT_SAMPLERATE); - LOGV("AudioStreamOutALSA::getPlaybackDelay delayNs: [%ld], "\ + ALOGV("AudioStreamOutALSA::getPlaybackDelay delayNs: [%ld], "\ "kernelFr:[%d], frames:[%d], buffSize:[%d], time_stamp:[%ld].[%ld]", delayNs, (int)kernelFr, (int)frames, pcm_get_buffer_size(mPcm), (long)buffer->time_stamp.tv_sec, buffer->time_stamp.tv_nsec); @@ -1325,7 +1324,7 @@ int AudioHardware::AudioStreamOutALSA::getPlaybackDelay(size_t frames, ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t bytes) { - //LOGV("-----AudioStreamInALSA::write(%p, %d) START", buffer, (int)bytes); + //ALOGV("-----AudioStreamInALSA::write(%p, %d) START", buffer, (int)bytes); status_t status = NO_INIT; const uint8_t* p = static_cast<const uint8_t*>(buffer); int ret; @@ -1345,7 +1344,7 @@ ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t byte if (mStandby) { AutoMutex hwLock(mHardware->lock()); - LOGD("AudioHardware pcm playback is exiting standby."); + ALOGD("AudioHardware pcm playback is exiting standby."); acquire_wake_lock(PARTIAL_WAKE_LOCK, "AudioOutLock"); sp<AudioStreamInALSA> spIn = mHardware->getActiveInput_l(); @@ -1359,7 +1358,7 @@ ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t byte // while the mutex is released if ((spIn == mHardware->getActiveInput_l()) && (cnt == spIn->standbyCnt())) { - LOGV("AudioStreamOutALSA::write() force input standby"); + ALOGV("AudioStreamOutALSA::write() force input standby"); spIn->close_l(); break; } @@ -1399,10 +1398,10 @@ ssize_t AudioHardware::AudioStreamOutALSA::write(const void* buffer, size_t byte TRACE_DRIVER_OUT if (ret == 0) { - //LOGV("-----AudioStreamInALSA::write(%p, %d) END", buffer, (int)bytes); + //ALOGV("-----AudioStreamInALSA::write(%p, %d) END", buffer, (int)bytes); return bytes; } - LOGW("write error: %d", errno); + ALOGW("write error: %d", errno); status = -errno; } Error: @@ -1410,7 +1409,7 @@ Error: // Simulate audio output timing in case of error usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate()); - LOGE("AudioStreamOutALSA::write END WITH ERROR !!!!!!!!!(%p, %u)", buffer, bytes); + ALOGE("AudioStreamOutALSA::write END WITH ERROR !!!!!!!!!(%p, %u)", buffer, bytes); return status; } @@ -1438,7 +1437,7 @@ void AudioHardware::AudioStreamOutALSA::doStandby_l() mStandbyCnt++; if (!mStandby) { - LOGD("AudioHardware pcm playback is going to standby."); + ALOGD("AudioHardware pcm playback is going to standby."); // stop echo reference capture if (mEchoReference != NULL) { mEchoReference->write(mEchoReference, NULL); @@ -1465,7 +1464,7 @@ void AudioHardware::AudioStreamOutALSA::close_l() status_t AudioHardware::AudioStreamOutALSA::open_l() { - LOGV("open pcm_out driver"); + ALOGV("open pcm_out driver"); mPcm = mHardware->openPcmOut_l(); if (mPcm == NULL) { return NO_INIT; @@ -1473,14 +1472,14 @@ status_t AudioHardware::AudioStreamOutALSA::open_l() mMixer = mHardware->openMixer_l(); if (mMixer) { - LOGV("open playback normal"); + ALOGV("open playback normal"); TRACE_DRIVER_IN(DRV_MIXER_GET) mRouteCtl = mixer_get_ctl_by_name(mMixer, "Playback Path"); TRACE_DRIVER_OUT } if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { const char *route = mHardware->getOutputRouteFromDevice(mDevices); - LOGV("write() wakeup setting route %s", route); + ALOGV("write() wakeup setting route %s", route); if (mRouteCtl) { TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_set_enum_by_string(mRouteCtl, route); @@ -1539,7 +1538,7 @@ status_t AudioHardware::AudioStreamOutALSA::setParameters(const String8& keyValu AudioParameter param = AudioParameter(keyValuePairs); status_t status = NO_ERROR; int device; - LOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string()); + ALOGD("AudioStreamOutALSA::setParameters() %s", keyValuePairs.string()); if (mHardware == NULL) return NO_INIT; @@ -1585,7 +1584,7 @@ String8 AudioHardware::AudioStreamOutALSA::getParameters(const String8& keys) param.addInt(key, (int)mDevices); } - LOGV("AudioStreamOutALSA::getParameters() %s", param.toString().string()); + ALOGV("AudioStreamOutALSA::getParameters() %s", param.toString().string()); return param.toString(); } @@ -1615,7 +1614,7 @@ void AudioHardware::AudioStreamOutALSA::unlock() { void AudioHardware::AudioStreamOutALSA::addEchoReference(struct echo_reference_itfe *reference) { - LOGV("AudioStreamOutALSA::addEchoReference %p", mEchoReference); + ALOGV("AudioStreamOutALSA::addEchoReference %p", mEchoReference); if (mEchoReference == NULL) { mEchoReference = reference; } @@ -1623,7 +1622,7 @@ void AudioHardware::AudioStreamOutALSA::addEchoReference(struct echo_reference_i void AudioHardware::AudioStreamOutALSA::removeEchoReference(struct echo_reference_itfe *reference) { - LOGV("AudioStreamOutALSA::removeEchoReference %p", mEchoReference); + ALOGV("AudioStreamOutALSA::removeEchoReference %p", mEchoReference); if (mEchoReference == reference) { mEchoReference->write(mEchoReference, NULL); mEchoReference = NULL; @@ -1671,7 +1670,7 @@ status_t AudioHardware::AudioStreamInALSA::set( mHardware = hw; - LOGV("AudioStreamInALSA::set(%d, %d, %u)", *pFormat, *pChannels, *pRate); + ALOGV("AudioStreamInALSA::set(%d, %d, %u)", *pFormat, *pChannels, *pRate); mBufferSize = getBufferSize(*pRate, AudioSystem::popCount(*pChannels)); mDevices = devices; @@ -1689,7 +1688,7 @@ status_t AudioHardware::AudioStreamInALSA::set( &mBufferProvider.mProvider, &mDownSampler); if (status != 0) { - LOGW("AudioStreamInALSA::set() downsampler init failed: %d", status); + ALOGW("AudioStreamInALSA::set() downsampler init failed: %d", status); mDownSampler = NULL; return status; } @@ -1758,7 +1757,7 @@ ssize_t AudioHardware::AudioStreamInALSA::processFrames(void* buffer, ssize_t fr mProcBufSize = (size_t)frames; mProcBuf = (int16_t *)realloc(mProcBuf, mProcBufSize * mChannelCount * sizeof(int16_t)); - LOGV("processFrames(): mProcBuf %p size extended to %d frames", + ALOGV("processFrames(): mProcBuf %p size extended to %d frames", mProcBuf, mProcBufSize); } ssize_t framesRd = readFrames(mProcBuf + mProcFramesIn * mChannelCount, @@ -1815,7 +1814,7 @@ int32_t AudioHardware::AudioStreamInALSA::updateEchoReference(size_t frames) struct echo_reference_buffer b; b.delay_ns = 0; - LOGV("updateEchoReference1 START, frames = [%d], mRefFramesIn = [%d], b.frame_count = [%d]", + ALOGV("updateEchoReference1 START, frames = [%d], mRefFramesIn = [%d], b.frame_count = [%d]", frames, mRefFramesIn, frames - mRefFramesIn); if (mRefFramesIn < frames) { if (mRefBufSize < frames) { @@ -1832,12 +1831,12 @@ int32_t AudioHardware::AudioStreamInALSA::updateEchoReference(size_t frames) if (mEchoReference->read(mEchoReference, &b) == NO_ERROR) { mRefFramesIn += b.frame_count; - LOGV("updateEchoReference2: mRefFramesIn:[%d], mRefBufSize:[%d], "\ + ALOGV("updateEchoReference2: mRefFramesIn:[%d], mRefBufSize:[%d], "\ "frames:[%d], b.frame_count:[%d]", mRefFramesIn, mRefBufSize,frames,b.frame_count); } }else{ - LOGV("updateEchoReference3: NOT enough frames to read ref buffer"); + ALOGV("updateEchoReference3: NOT enough frames to read ref buffer"); } return b.delay_ns; } @@ -1869,7 +1868,7 @@ void AudioHardware::AudioStreamInALSA::pushEchoReference(size_t frames) mRefFramesIn -= refBuf.frameCount; if (mRefFramesIn) { - LOGV("pushEchoReference5: shifting mRefBuf down by = %d frames", mRefFramesIn); + ALOGV("pushEchoReference5: shifting mRefBuf down by = %d frames", mRefFramesIn); memcpy(mRefBuf, mRefBuf + refBuf.frameCount * mChannelCount, mRefFramesIn * mChannelCount * sizeof(int16_t)); @@ -1887,7 +1886,7 @@ status_t AudioHardware::AudioStreamInALSA::setPreProcessorEchoDelay(effect_handl *(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY; *((int32_t *)param->data + 1) = delayUs; - LOGV("setPreProcessorEchoDelay: %d us", delayUs); + ALOGV("setPreProcessorEchoDelay: %d us", delayUs); return setPreprocessorParam(handle, param); } @@ -1922,7 +1921,7 @@ void AudioHardware::AudioStreamInALSA::getCaptureDelay(size_t frames, buffer->time_stamp.tv_sec = 0; buffer->time_stamp.tv_nsec = 0; buffer->delay_ns = 0; - LOGW("read getCaptureDelay(): pcm_htimestamp error"); + ALOGW("read getCaptureDelay(): pcm_htimestamp error"); return; } @@ -1944,7 +1943,7 @@ void AudioHardware::AudioStreamInALSA::getCaptureDelay(size_t frames, buffer->time_stamp = tstamp; buffer->delay_ns = delayNs; - LOGV("AudioStreamInALSA::getCaptureDelay TimeStamp = [%ld].[%ld], delayCaptureNs: [%d],"\ + ALOGV("AudioStreamInALSA::getCaptureDelay TimeStamp = [%ld].[%ld], delayCaptureNs: [%d],"\ " kernelDelay:[%ld], bufDelay:[%ld], rsmpDelay:[%ld], kernelFr:[%d], "\ "mInputFramesIn:[%d], mProcFramesIn:[%d], frames:[%d]", buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns, @@ -1954,7 +1953,7 @@ void AudioHardware::AudioStreamInALSA::getCaptureDelay(size_t frames, ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes) { - //LOGV("-----AudioStreamInALSA::read(%p, %d) START", buffer, (int)bytes); + //ALOGV("-----AudioStreamInALSA::read(%p, %d) START", buffer, (int)bytes); status_t status = NO_INIT; if (mHardware == NULL) return NO_INIT; @@ -1971,7 +1970,7 @@ ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes) if (mStandby) { AutoMutex hwLock(mHardware->lock()); - LOGD("AudioHardware pcm capture is exiting standby."); + ALOGD("AudioHardware pcm capture is exiting standby."); sp<AudioStreamOutALSA> spOut = mHardware->output(); while (spOut != 0) { spOut->prepareLock(); @@ -1992,13 +1991,13 @@ ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes) // open output before input if (spOut != 0) { if (!spOut->checkStandby()) { - LOGV("AudioStreamInALSA::read() force output standby"); + ALOGV("AudioStreamInALSA::read() force output standby"); spOut->close_l(); if (spOut->open_l() != NO_ERROR) { spOut->doStandby_l(); } } - LOGV("AudioStreamInALSA exit standby mNeedEchoReference %d mEchoReference %p", + ALOGV("AudioStreamInALSA exit standby mNeedEchoReference %d mEchoReference %p", mNeedEchoReference, mEchoReference); if (mNeedEchoReference && mEchoReference == NULL) { mEchoReference = mHardware->getEchoReference(AUDIO_FORMAT_PCM_16_BIT, @@ -2026,11 +2025,11 @@ ssize_t AudioHardware::AudioStreamInALSA::read(void* buffer, ssize_t bytes) } if (framesRd >= 0) { - //LOGV("-----AudioStreamInALSA::read(%p, %d) END", buffer, (int)bytes); + //ALOGV("-----AudioStreamInALSA::read(%p, %d) END", buffer, (int)bytes); return framesRd * mChannelCount * sizeof(int16_t); } - LOGW("read error: %d", (int)framesRd); + ALOGW("read error: %d", (int)framesRd); status = framesRd; } @@ -2040,7 +2039,7 @@ Error: // Simulate audio output timing in case of error usleep((((bytes * 1000) / frameSize()) * 1000) / sampleRate()); - LOGE("-----AudioStreamInALSA::read(%p, %d) END ERROR", buffer, (int)bytes); + ALOGE("-----AudioStreamInALSA::read(%p, %d) END ERROR", buffer, (int)bytes); return status; } @@ -2067,7 +2066,7 @@ void AudioHardware::AudioStreamInALSA::doStandby_l() mStandbyCnt++; if (!mStandby) { - LOGD("AudioHardware pcm capture is going to standby."); + ALOGD("AudioHardware pcm capture is going to standby."); if (mEchoReference != NULL) { // stop reading from echo reference mEchoReference->read(mEchoReference, NULL); @@ -2129,12 +2128,12 @@ status_t AudioHardware::AudioStreamInALSA::open_l() silence_threshold : 0, }; - LOGV("open pcm_in driver"); + ALOGV("open pcm_in driver"); TRACE_DRIVER_IN(DRV_PCM_OPEN) mPcm = pcm_open(0, 0, flags, &config); TRACE_DRIVER_OUT if (!pcm_is_ready(mPcm)) { - LOGE("cannot open pcm_in driver: %s\n", pcm_get_error(mPcm)); + ALOGE("cannot open pcm_in driver: %s\n", pcm_get_error(mPcm)); TRACE_DRIVER_IN(DRV_PCM_CLOSE) pcm_close(mPcm); TRACE_DRIVER_OUT @@ -2161,7 +2160,7 @@ status_t AudioHardware::AudioStreamInALSA::open_l() if (mHardware->mode() != AudioSystem::MODE_IN_CALL) { const char *route = mHardware->getInputRouteFromDevice(mDevices); - LOGV("read() wakeup setting route %s", route); + ALOGV("read() wakeup setting route %s", route); if (mRouteCtl) { TRACE_DRIVER_IN(DRV_MIXER_SEL) mixer_ctl_set_enum_by_string(mRouteCtl, route); @@ -2219,7 +2218,7 @@ status_t AudioHardware::AudioStreamInALSA::setParameters(const String8& keyValue status_t status = NO_ERROR; int value; - LOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); + ALOGD("AudioStreamInALSA::setParameters() %s", keyValuePairs.string()); if (mHardware == NULL) return NO_INIT; @@ -2273,25 +2272,25 @@ String8 AudioHardware::AudioStreamInALSA::getParameters(const String8& keys) param.addInt(key, (int)mDevices); } - LOGV("AudioStreamInALSA::getParameters() %s", param.toString().string()); + ALOGV("AudioStreamInALSA::getParameters() %s", param.toString().string()); return param.toString(); } status_t AudioHardware::AudioStreamInALSA::addAudioEffect(effect_handle_t effect) { - LOGV("AudioStreamInALSA::addAudioEffect() %p", effect); + ALOGV("AudioStreamInALSA::addAudioEffect() %p", effect); effect_descriptor_t desc; status_t status = (*effect)->get_descriptor(effect, &desc); if (status == 0) { if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { - LOGV("AudioStreamInALSA::addAudioEffect() mNeedEchoReference true"); + ALOGV("AudioStreamInALSA::addAudioEffect() mNeedEchoReference true"); mNeedEchoReference = true; standby(); } - LOGV("AudioStreamInALSA::addAudioEffect() name %s", desc.name); + ALOGV("AudioStreamInALSA::addAudioEffect() name %s", desc.name); } else { - LOGV("AudioStreamInALSA::addAudioEffect() get_descriptor() error"); + ALOGV("AudioStreamInALSA::addAudioEffect() get_descriptor() error"); } AutoMutex lock(mLock); @@ -2302,7 +2301,7 @@ status_t AudioHardware::AudioStreamInALSA::addAudioEffect(effect_handle_t effect status_t AudioHardware::AudioStreamInALSA::removeAudioEffect(effect_handle_t effect) { status_t status = INVALID_OPERATION; - LOGV("AudioStreamInALSA::removeAudioEffect() %p", effect); + ALOGV("AudioStreamInALSA::removeAudioEffect() %p", effect); { AutoMutex lock(mLock); for (size_t i = 0; i < mPreprocessors.size(); i++) { @@ -2318,7 +2317,7 @@ status_t AudioHardware::AudioStreamInALSA::removeAudioEffect(effect_handle_t eff effect_descriptor_t desc; if ((*effect)->get_descriptor(effect, &desc) == 0) { if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { - LOGV("AudioStreamInALSA::removeAudioEffect() mNeedEchoReference false"); + ALOGV("AudioStreamInALSA::removeAudioEffect() mNeedEchoReference false"); mNeedEchoReference = false; standby(); } @@ -2392,7 +2391,7 @@ size_t AudioHardware::AudioStreamInALSA::getBufferSize(uint32_t sampleRate, int } // this should never happen as getBufferSize() is always called after getInputSampleRate() // that checks for valid sampling rates. - LOGE("AudioStreamInALSA::getBufferSize() invalid sampling rate %d", sampleRate); + ALOGE("AudioStreamInALSA::getBufferSize() invalid sampling rate %d", sampleRate); return 0; } diff --git a/libaudio/AudioHardware.h b/libaudio/AudioHardware.h index 2fcf4f3..4d2e2eb 100644 --- a/libaudio/AudioHardware.h +++ b/libaudio/AudioHardware.h @@ -24,7 +24,6 @@ #include <utils/SortedVector.h> #include <hardware_legacy/AudioHardwareBase.h> -#include <media/mediarecorder.h> #include <hardware/audio_effect.h> #include "secril-client.h" @@ -48,7 +47,7 @@ namespace android_audio_legacy { using android::Vector; // TODO: determine actual audio DSP and hardware latency -// Additionnal latency introduced by audio DSP and hardware in ms +// Additional latency introduced by audio DSP and hardware in ms #define AUDIO_HW_OUT_LATENCY_MS 0 // Default audio output sample rate #define AUDIO_HW_OUT_SAMPLERATE 44100 @@ -57,8 +56,8 @@ namespace android_audio_legacy { // Default audio output sample format #define AUDIO_HW_OUT_FORMAT (AudioSystem::PCM_16_BIT) // Kernel pcm out buffer size in frames at 44.1kHz -#define AUDIO_HW_OUT_PERIOD_SZ 1024 -#define AUDIO_HW_OUT_PERIOD_CNT 4 +#define AUDIO_HW_OUT_PERIOD_SZ 880 +#define AUDIO_HW_OUT_PERIOD_CNT 2 // Default audio output buffer size in bytes #define AUDIO_HW_OUT_PERIOD_BYTES (AUDIO_HW_OUT_PERIOD_SZ * 2 * sizeof(int16_t)) diff --git a/libaudio/AudioPolicyManager.cpp b/libaudio/AudioPolicyManager.cpp index bfa6c00..ce99972 100644 --- a/libaudio/AudioPolicyManager.cpp +++ b/libaudio/AudioPolicyManager.cpp @@ -18,7 +18,6 @@ //#define LOG_NDEBUG 0 #include <utils/Log.h> #include "AudioPolicyManager.h" -#include <media/mediarecorder.h> namespace android_audio_legacy { diff --git a/libaudio/AudioPolicyManager.h b/libaudio/AudioPolicyManager.h index d660e78..5062ff0 100644 --- a/libaudio/AudioPolicyManager.h +++ b/libaudio/AudioPolicyManager.h @@ -34,13 +34,5 @@ public: virtual ~AudioPolicyManager() {} -protected: - // true is current platform implements a back microphone - virtual bool hasBackMicrophone() const { return false; } -#ifdef WITH_A2DP - // true is current platform supports duplication of notifications and ringtones over A2DP output - virtual bool a2dpUsedForSonification() const { return true; } -#endif - }; }; diff --git a/libaudio/audio_policy.conf b/libaudio/audio_policy.conf new file mode 100644 index 0000000..9e980a0 --- /dev/null +++ b/libaudio/audio_policy.conf @@ -0,0 +1,68 @@ +# Global configuration section: lists input and output devices always present on the device +# as well as the output device selected by default. +# Devices are designated by a string that corresponds to the enum in audio.h + +global_configuration { + attached_output_devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER + default_output_device AUDIO_DEVICE_OUT_SPEAKER + attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC +} + +# audio hardware module section: contains descriptors for all audio hw modules present on the +# device. Each hw module node is named after the corresponding hw module library base name. +# For instance, "primary" corresponds to audio.primary.<device>.so. +# The "primary" module is mandatory and must include at least one output with +# AUDIO_OUTPUT_FLAG_PRIMARY flag. +# Each module descriptor contains one or more output profile descriptors and zero or more +# input profile descriptors. Each profile lists all the parameters supported by a given output +# or input stream category. +# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding +# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "\n". + +audio_hw_modules { + primary { + outputs { + primary { + sampling_rates 44100 + channel_masks AUDIO_CHANNEL_OUT_STEREO + formats AUDIO_FORMAT_PCM_16_BIT + devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_ALL_SCO + flags AUDIO_OUTPUT_FLAG_PRIMARY + } + } + inputs { + primary { + sampling_rates 8000|11025|16000|22050|32000|44100 + channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO + formats AUDIO_FORMAT_PCM_16_BIT + devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_WIRED_HEADSET + } + } + } + a2dp { + outputs { + a2dp { + sampling_rates 44100 + channel_masks AUDIO_CHANNEL_OUT_STEREO + formats AUDIO_FORMAT_PCM_16_BIT + devices AUDIO_DEVICE_OUT_ALL_A2DP + } + } + } + usb { + outputs { + usb_accessory { + sampling_rates 44100 + channel_masks AUDIO_CHANNEL_OUT_STEREO + formats AUDIO_FORMAT_PCM_16_BIT + devices AUDIO_DEVICE_OUT_USB_ACCESSORY + } + usb_device { + sampling_rates 44100 + channel_masks AUDIO_CHANNEL_OUT_STEREO + formats AUDIO_FORMAT_PCM_16_BIT + devices AUDIO_DEVICE_OUT_USB_DEVICE + } + } + } +} |