initial audio HAL implementation for mako
alsa_sound is imported from codeaurora at:
c1217338f349fe746e0933fcf9b1b288b532808d
[remote "quic"]
url = git://git-android.quicinc.com/platform/hardware/alsa_sound.git
review = review-android.quicinc.com
projectname = platform/hardware/alsa_sound
fetch = +refs/heads/*:refs/remotes/quic/*
Change-Id: Ic985cc3a1088c3957b6e2ac5537e2c36caaf7212
Signed-off-by: Iliyan Malchev <malchev@google.com>
diff --git a/alsa_sound/ALSAStreamOps.cpp b/alsa_sound/ALSAStreamOps.cpp
new file mode 100644
index 0000000..bbb556c
--- /dev/null
+++ b/alsa_sound/ALSAStreamOps.cpp
@@ -0,0 +1,360 @@
+/* ALSAStreamOps.cpp
+ **
+ ** Copyright 2008-2009 Wind River Systems
+ ** Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ **
+ ** 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 <sys/stat.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <dlfcn.h>
+
+#define LOG_TAG "ALSAStreamOps"
+//#define LOG_NDEBUG 0
+#define LOG_NDDEBUG 0
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include <cutils/properties.h>
+#include <media/AudioRecord.h>
+#include <hardware_legacy/power.h>
+
+#include "AudioHardwareALSA.h"
+
+namespace android_audio_legacy
+{
+
+// ----------------------------------------------------------------------------
+
+ALSAStreamOps::ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle) :
+ mParent(parent),
+ mHandle(handle)
+{
+}
+
+ALSAStreamOps::~ALSAStreamOps()
+{
+ Mutex::Autolock autoLock(mParent->mLock);
+
+ if((!strcmp(mHandle->useCase, SND_USE_CASE_VERB_IP_VOICECALL)) ||
+ (!strcmp(mHandle->useCase, SND_USE_CASE_MOD_PLAY_VOIP))) {
+ if((mParent->mVoipStreamCount)) {
+ mParent->mVoipStreamCount--;
+ if(mParent->mVoipStreamCount > 0) {
+ LOGD("ALSAStreamOps::close() Ignore");
+ return ;
+ }
+ }
+ mParent->mVoipStreamCount = 0;
+ mParent->mVoipMicMute = 0;
+ mParent->mVoipBitRate = 0;
+ }
+ close();
+
+ for(ALSAHandleList::iterator it = mParent->mDeviceList.begin();
+ it != mParent->mDeviceList.end(); ++it) {
+ if (mHandle == &(*it)) {
+ it->useCase[0] = 0;
+ mParent->mDeviceList.erase(it);
+ break;
+ }
+ }
+}
+
+// use emulated popcount optimization
+// http://www.df.lth.se/~john_e/gems/gem002d.html
+static inline uint32_t popCount(uint32_t u)
+{
+ u = ((u&0x55555555) + ((u>>1)&0x55555555));
+ u = ((u&0x33333333) + ((u>>2)&0x33333333));
+ u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
+ u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
+ u = ( u&0x0000ffff) + (u>>16);
+ return u;
+}
+
+status_t ALSAStreamOps::set(int *format,
+ uint32_t *channels,
+ uint32_t *rate,
+ uint32_t device)
+{
+ mDevices = device;
+ if (channels && *channels != 0) {
+ if (mHandle->channels != popCount(*channels))
+ return BAD_VALUE;
+ } else if (channels) {
+ *channels = 0;
+ if (mHandle->devices & AudioSystem::DEVICE_OUT_ALL) {
+ switch(mHandle->channels) {
+ case 4:
+ *channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
+ *channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
+ // Fall through...
+ default:
+ case 2:
+ *channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
+ // Fall through...
+ case 1:
+ *channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
+ break;
+ }
+ } else {
+ switch(mHandle->channels) {
+#ifdef SSR_ENABLED
+ // For 5.1 recording
+ case 6 :
+ *channels |= AudioSystem::CHANNEL_IN_5POINT1;
+ break;
+#endif
+ // Do not fall through...
+ default:
+ case 2:
+ *channels |= AudioSystem::CHANNEL_IN_RIGHT;
+ // Fall through...
+ case 1:
+ *channels |= AudioSystem::CHANNEL_IN_LEFT;
+ break;
+ }
+ }
+ }
+
+ if (rate && *rate > 0) {
+ if (mHandle->sampleRate != *rate)
+ return BAD_VALUE;
+ } else if (rate) {
+ *rate = mHandle->sampleRate;
+ }
+
+ snd_pcm_format_t iformat = mHandle->format;
+
+ if (format) {
+ switch(*format) {
+ case AudioSystem::FORMAT_DEFAULT:
+ break;
+
+ case AudioSystem::PCM_16_BIT:
+ iformat = SNDRV_PCM_FORMAT_S16_LE;
+ break;
+ case AudioSystem::AMR_NB:
+ case AudioSystem::AMR_WB:
+#if 0
+ case AudioSystem::EVRC:
+ case AudioSystem::EVRCB:
+ case AudioSystem::EVRCWB:
+#endif
+ iformat = *format;
+ break;
+
+ case AudioSystem::PCM_8_BIT:
+ iformat = SNDRV_PCM_FORMAT_S8;
+ break;
+
+ default:
+ LOGE("Unknown PCM format %i. Forcing default", *format);
+ break;
+ }
+
+ if (mHandle->format != iformat)
+ return BAD_VALUE;
+
+ switch(iformat) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ *format = AudioSystem::PCM_16_BIT;
+ break;
+ case SNDRV_PCM_FORMAT_S8:
+ *format = AudioSystem::PCM_8_BIT;
+ break;
+ default:
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t ALSAStreamOps::setParameters(const String8& keyValuePairs)
+{
+ AudioParameter param = AudioParameter(keyValuePairs);
+ String8 key = String8(AudioParameter::keyRouting);
+ int device;
+ if (param.getInt(key, device) == NO_ERROR) {
+ // Ignore routing if device is 0.
+ LOGD("setParameters(): keyRouting with device %d", device);
+ mDevices = device;
+ if(device) {
+ mParent->doRouting(device);
+ }
+ param.remove(key);
+ }
+#ifdef FM_ENABLED
+ else {
+ key = String8(AudioParameter::keyHandleFm);
+ if (param.getInt(key, device) == NO_ERROR) {
+ LOGD("setParameters(): handleFm with device %d", device);
+ mDevices = device;
+ if(device) {
+ mParent->handleFm(device);
+ }
+ param.remove(key);
+ }
+ }
+#endif
+
+ return NO_ERROR;
+}
+
+String8 ALSAStreamOps::getParameters(const String8& keys)
+{
+ AudioParameter param = AudioParameter(keys);
+ String8 value;
+ String8 key = String8(AudioParameter::keyRouting);
+
+ if (param.get(key, value) == NO_ERROR) {
+ param.addInt(key, (int)mDevices);
+ }
+ else {
+ key = String8(AudioParameter::keyVoipCheck);
+ if (param.get(key, value) == NO_ERROR) {
+ if((!strncmp(mHandle->useCase, SND_USE_CASE_VERB_IP_VOICECALL, strlen(SND_USE_CASE_VERB_IP_VOICECALL))) ||
+ (!strncmp(mHandle->useCase, SND_USE_CASE_MOD_PLAY_VOIP, strlen(SND_USE_CASE_MOD_PLAY_VOIP))))
+ param.addInt(key, true);
+ else
+ param.addInt(key, false);
+ }
+ }
+ LOGV("getParameters() %s", param.toString().string());
+ return param.toString();
+}
+
+uint32_t ALSAStreamOps::sampleRate() const
+{
+ return mHandle->sampleRate;
+}
+
+//
+// Return the number of bytes (not frames)
+//
+size_t ALSAStreamOps::bufferSize() const
+{
+ LOGV("bufferSize() returns %d", mHandle->bufferSize);
+ return mHandle->bufferSize;
+}
+
+int ALSAStreamOps::format() const
+{
+ int audioSystemFormat;
+
+ snd_pcm_format_t ALSAFormat = mHandle->format;
+
+ switch(ALSAFormat) {
+ case SNDRV_PCM_FORMAT_S8:
+ audioSystemFormat = AudioSystem::PCM_8_BIT;
+ break;
+
+ case AudioSystem::AMR_NB:
+ case AudioSystem::AMR_WB:
+#if 0
+ case AudioSystem::EVRC:
+ case AudioSystem::EVRCB:
+ case AudioSystem::EVRCWB:
+#endif
+ audioSystemFormat = mHandle->format;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ audioSystemFormat = AudioSystem::PCM_16_BIT;
+ break;
+
+ default:
+ LOG_FATAL("Unknown AudioSystem bit width %d!", audioSystemFormat);
+ audioSystemFormat = AudioSystem::PCM_16_BIT;
+ break;
+ }
+
+ LOGD("ALSAFormat:0x%x,audioSystemFormat:0x%x",ALSAFormat,audioSystemFormat);
+ return audioSystemFormat;
+}
+
+uint32_t ALSAStreamOps::channels() const
+{
+ unsigned int count = mHandle->channels;
+ uint32_t channels = 0;
+
+ if (mDevices & AudioSystem::DEVICE_OUT_ALL)
+ switch(count) {
+ case 4:
+ channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
+ channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
+ // Fall through...
+ default:
+ case 2:
+ channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
+ // Fall through...
+ case 1:
+ channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
+ break;
+ }
+ else
+ switch(count) {
+#ifdef SSR_ENABLED
+ // For 5.1 recording
+ case 6 :
+ channels |= AudioSystem::CHANNEL_IN_5POINT1;
+ break;
+ // Do not fall through...
+#endif
+ default:
+ case 2:
+ channels |= AudioSystem::CHANNEL_IN_RIGHT;
+ // Fall through...
+ case 1:
+ channels |= AudioSystem::CHANNEL_IN_LEFT;
+ break;
+ }
+
+ return channels;
+}
+
+void ALSAStreamOps::close()
+{
+ LOGD("close");
+ if((!strncmp(mHandle->useCase, SND_USE_CASE_VERB_IP_VOICECALL, strlen(SND_USE_CASE_VERB_IP_VOICECALL))) ||
+ (!strncmp(mHandle->useCase, SND_USE_CASE_MOD_PLAY_VOIP, strlen(SND_USE_CASE_MOD_PLAY_VOIP)))) {
+ mParent->mVoipMicMute = false;
+ mParent->mVoipBitRate = 0;
+ mParent->mVoipStreamCount = 0;
+ }
+ mParent->mALSADevice->close(mHandle);
+}
+
+//
+// Set playback or capture PCM device. It's possible to support audio output
+// or input from multiple devices by using the ALSA plugins, but this is
+// not supported for simplicity.
+//
+// The AudioHardwareALSA API does not allow one to set the input routing.
+//
+// If the "routes" value does not map to a valid device, the default playback
+// device is used.
+//
+status_t ALSAStreamOps::open(int mode)
+{
+ LOGD("open");
+ return mParent->mALSADevice->open(mHandle);
+}
+
+} // namespace androidi_audio_legacy