Template files from frameworks/base/libs/audioflinger

3 files changed, 510 insertions, 0 deletions

diff --git a/AudioHardwareInterface.cpp b/AudioHardwareInterface.cpp
new file mode 100644
index 0000000..7387b3d
--- /dev/null
+++ b/AudioHardwareInterface.cpp
@@ -0,0 +1,240 @@
+/*
+**
+** Copyright 2007, The Android Open Source Project
+**
+** 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 <cutils/properties.h>
+#include <string.h>
+#include <unistd.h>
+
+#define LOG_TAG "AudioHardwareInterface"
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include "AudioHardwareStub.h"
+#include "AudioHardwareGeneric.h"
+
+// #define DUMP_FLINGER_OUT        // if defined allows recording samples in a file
+#ifdef DUMP_FLINGER_OUT
+#include "AudioDumpInterface.h"
+#endif
+
+
+// change to 1 to log routing calls
+#define LOG_ROUTING_CALLS 0
+
+namespace android {
+
+#if LOG_ROUTING_CALLS
+static const char* routingModeStrings[] =
+{
+    "OUT OF RANGE",
+    "INVALID",
+    "CURRENT",
+    "NORMAL",
+    "RINGTONE",
+    "IN_CALL"
+};
+
+static const char* routeStrings[] =
+{
+    "EARPIECE ",
+    "SPEAKER ",
+    "BLUETOOTH ",
+    "HEADSET "
+};
+static const char* routeNone = "NONE";
+
+static const char* displayMode(int mode)
+{
+    if ((mode < -2) || (mode > 2))
+        return routingModeStrings[0];
+    return routingModeStrings[mode+3];
+}
+
+static const char* displayRoutes(uint32_t routes)
+{
+    static char routeStr[80];
+    if (routes == 0)
+        return routeNone;
+    routeStr[0] = 0;
+    int bitMask = 1;
+    for (int i = 0; i < 4; ++i, bitMask <<= 1) {
+        if (routes & bitMask) {
+            strcat(routeStr, routeStrings[i]);
+        }
+    }
+    routeStr[strlen(routeStr)-1] = 0;
+    return routeStr;
+}
+#endif
+
+// ----------------------------------------------------------------------------
+
+AudioHardwareInterface* AudioHardwareInterface::create()
+{
+    /*
+     * FIXME: This code needs to instantiate the correct audio device
+     * interface. For now - we use compile-time switches.
+     */
+    AudioHardwareInterface* hw = 0;
+    char value[PROPERTY_VALUE_MAX];
+
+#ifdef GENERIC_AUDIO
+    hw = new AudioHardwareGeneric();
+#else
+    // if running in emulation - use the emulator driver
+    if (property_get("ro.kernel.qemu", value, 0)) {
+        LOGD("Running in emulation - using generic audio driver");
+        hw = new AudioHardwareGeneric();
+    }
+    else {
+        LOGV("Creating Vendor Specific AudioHardware");
+        hw = createAudioHardware();
+    }
+#endif
+    if (hw->initCheck() != NO_ERROR) {
+        LOGW("Using stubbed audio hardware. No sound will be produced.");
+        delete hw;
+        hw = new AudioHardwareStub();
+    }
+    
+#ifdef DUMP_FLINGER_OUT
+    // This code adds a record of buffers in a file to write calls made by AudioFlinger.
+    // It replaces the current AudioHardwareInterface object by an intermediate one which
+    // will record buffers in a file (after sending them to hardware) for testing purpose.
+    // This feature is enabled by defining symbol DUMP_FLINGER_OUT and setting environement
+    // "audioflinger.dump = 1". The output file is "tmp/FlingerOut.pcm". Pause are not recorded
+    // in the file.
+    
+    // read dump mode
+    property_get("audioflinger.dump", value, "0");
+    switch(value[0]) {
+    case '1':
+        LOGV("Dump mode");
+        hw = new AudioDumpInterface(hw);    // replace interface
+        return hw;
+        break;
+    case '0':
+    default:
+        LOGV("No Dump mode");
+        return hw;
+        break;
+    }
+#endif
+    return hw;
+}
+
+AudioStreamOut::~AudioStreamOut()
+{
+}
+
+AudioStreamIn::~AudioStreamIn() {}
+
+AudioHardwareInterface::AudioHardwareInterface()
+{
+    // force a routing update on initialization
+    memset(&mRoutes, 0, sizeof(mRoutes));
+    mMode = 0;
+}
+
+// generics for audio routing - the real work is done in doRouting
+status_t AudioHardwareInterface::setRouting(int mode, uint32_t routes)
+{
+#if LOG_ROUTING_CALLS
+    LOGD("setRouting: mode=%s, routes=[%s]", displayMode(mode), displayRoutes(routes));
+#endif
+    if (mode == AudioSystem::MODE_CURRENT)
+        mode = mMode;
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    uint32_t old = mRoutes[mode];
+    mRoutes[mode] = routes;
+    if ((mode != mMode) || (old == routes))
+        return NO_ERROR;
+#if LOG_ROUTING_CALLS
+    const char* oldRouteStr = strdup(displayRoutes(old));
+    LOGD("doRouting: mode=%s, old route=[%s], new route=[%s]",
+           displayMode(mode), oldRouteStr, displayRoutes(routes));
+    delete oldRouteStr;
+#endif
+    return doRouting();
+}
+
+status_t AudioHardwareInterface::getRouting(int mode, uint32_t* routes)
+{
+    if (mode == AudioSystem::MODE_CURRENT)
+        mode = mMode;
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    *routes = mRoutes[mode];
+#if LOG_ROUTING_CALLS
+    LOGD("getRouting: mode=%s, routes=[%s]",
+           displayMode(mode), displayRoutes(*routes));
+#endif
+    return NO_ERROR;
+}
+
+status_t AudioHardwareInterface::setMode(int mode)
+{
+#if LOG_ROUTING_CALLS
+    LOGD("setMode(%s)", displayMode(mode));
+#endif
+    if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
+        return BAD_VALUE;
+    if (mMode == mode)
+        return NO_ERROR;
+#if LOG_ROUTING_CALLS
+    LOGD("doRouting: old mode=%s, new mode=%s route=[%s]",
+            displayMode(mMode), displayMode(mode), displayRoutes(mRoutes[mode]));
+#endif
+    mMode = mode;
+    return doRouting();
+}
+
+status_t AudioHardwareInterface::getMode(int* mode)
+{
+    // Implement: set audio routing
+    *mode = mMode;
+    return NO_ERROR;
+}
+
+status_t AudioHardwareInterface::setParameter(const char* key, const char* value)
+{
+    // default implementation is to ignore
+    return NO_ERROR;
+}
+
+status_t AudioHardwareInterface::dumpState(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "AudioHardwareInterface::dumpState\n");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
+    result.append(buffer);
+    for (int i = 0, n = AudioSystem::NUM_MODES; i < n; ++i) {
+        snprintf(buffer, SIZE, "\tmRoutes[%d]: %d\n", i, mRoutes[i]);
+        result.append(buffer);
+    }
+    ::write(fd, result.string(), result.size());
+    dump(fd, args);  // Dump the state of the concrete child.
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/AudioHardwareStub.cpp b/AudioHardwareStub.cpp
new file mode 100644
index 0000000..0046db8
--- /dev/null
+++ b/AudioHardwareStub.cpp
@@ -0,0 +1,175 @@
+/* //device/servers/AudioFlinger/AudioHardwareStub.cpp
+**
+** Copyright 2007, The Android Open Source Project
+**
+** 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 <stdint.h>
+#include <sys/types.h>
+
+#include <stdlib.h>
+#include <unistd.h>
+#include <utils/String8.h>
+
+#include "AudioHardwareStub.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+AudioHardwareStub::AudioHardwareStub() : mMicMute(false)
+{
+}
+
+AudioHardwareStub::~AudioHardwareStub()
+{
+}
+
+status_t AudioHardwareStub::initCheck()
+{
+    return NO_ERROR;
+}
+
+status_t AudioHardwareStub::standby()
+{
+    return NO_ERROR;
+}
+
+AudioStreamOut* AudioHardwareStub::openOutputStream(
+        int format, int channelCount, uint32_t sampleRate)
+{
+    AudioStreamOutStub* out = new AudioStreamOutStub();
+    if (out->set(format, channelCount, sampleRate) == NO_ERROR)
+        return out;
+    delete out;
+    return 0;
+}
+
+AudioStreamIn* AudioHardwareStub::openInputStream(
+        int format, int channelCount, uint32_t sampleRate)
+{
+    AudioStreamInStub* in = new AudioStreamInStub();
+    if (in->set(format, channelCount, sampleRate) == NO_ERROR)
+        return in;
+    delete in;
+    return 0;
+}
+
+status_t AudioHardwareStub::setVoiceVolume(float volume)
+{
+    return NO_ERROR;
+}
+
+status_t AudioHardwareStub::setMasterVolume(float volume)
+{
+    return NO_ERROR;
+}
+
+status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    result.append("AudioHardwareStub::dumpInternals\n");
+    snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
+{
+    dumpInternals(fd, args);
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+
+status_t AudioStreamOutStub::set(int format, int channels, uint32_t rate)
+{
+    // fix up defaults
+    if (format == 0) format = AudioSystem::PCM_16_BIT;
+    if (channels == 0) channels = channelCount();
+    if (rate == 0) rate = sampleRate();
+    
+    if ((format == AudioSystem::PCM_16_BIT) &&
+            (channels == channelCount()) &&
+            (rate == sampleRate()))
+        return NO_ERROR;
+    return BAD_VALUE;
+}
+
+ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
+{
+    // fake timing for audio output
+    usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+    return bytes;
+}
+
+status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
+    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
+    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
+    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    snprintf(buffer, SIZE, "\tformat: %d\n", format());
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR; 
+}
+
+// ----------------------------------------------------------------------------
+
+status_t AudioStreamInStub::set(int format, int channels, uint32_t rate)
+{
+    if ((format == AudioSystem::PCM_16_BIT) &&
+            (channels == channelCount()) &&
+            (rate == sampleRate()))
+        return NO_ERROR;
+    return BAD_VALUE;
+}
+
+ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
+{
+    // fake timing for audio input
+    usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate());
+    memset(buffer, 0, bytes);
+    return bytes;
+}
+
+status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+    snprintf(buffer, SIZE, "AudioStreamInStub::dump\n");
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tchannel count: %d\n", channelCount());
+    result.append(buffer);
+    snprintf(buffer, SIZE, "\tformat: %d\n", format());
+    result.append(buffer);
+    ::write(fd, result.string(), result.size());
+    return NO_ERROR;
+}
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
diff --git a/AudioHardwareStub.h b/AudioHardwareStub.h
new file mode 100644
index 0000000..1a61552
--- /dev/null
+++ b/AudioHardwareStub.h
@@ -0,0 +1,95 @@
+/* //device/servers/AudioFlinger/AudioHardwareStub.h
+**
+** Copyright 2007, The Android Open Source Project
+**
+** 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_STUB_H
+#define ANDROID_AUDIO_HARDWARE_STUB_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <hardware/AudioHardwareInterface.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioStreamOutStub : public AudioStreamOut {
+public:
+    virtual status_t    set(int format, int channelCount, uint32_t sampleRate);
+    virtual uint32_t    sampleRate() const { return 44100; }
+    virtual size_t      bufferSize() const { return 4096; }
+    virtual int         channelCount() const { return 2; }
+    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
+    virtual status_t    setVolume(float volume) { return NO_ERROR; }
+    virtual ssize_t     write(const void* buffer, size_t bytes);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+};
+
+class AudioStreamInStub : public AudioStreamIn {
+public:
+    virtual status_t    set(int format, int channelCount, uint32_t sampleRate);
+    virtual uint32_t    sampleRate() const { return 8000; }
+    virtual size_t      bufferSize() const { return 320; }
+    virtual int         channelCount() const { return 1; }
+    virtual int         format() const { return AudioSystem::PCM_16_BIT; }
+    virtual status_t    setGain(float gain) { return NO_ERROR; }
+    virtual ssize_t     read(void* buffer, ssize_t bytes);
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+};
+
+class AudioHardwareStub : public  AudioHardwareInterface
+{
+public:
+                        AudioHardwareStub();
+    virtual             ~AudioHardwareStub();
+    virtual status_t    initCheck();
+    virtual status_t    standby();
+    virtual status_t    setVoiceVolume(float volume);
+    virtual status_t    setMasterVolume(float volume);
+
+    // mic mute
+    virtual status_t    setMicMute(bool state) { mMicMute = state;  return  NO_ERROR; }
+    virtual status_t    getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; }
+
+    virtual status_t    setParameter(const char* key, const char* value)
+            { return NO_ERROR; }
+
+    // create I/O streams
+    virtual AudioStreamOut* openOutputStream(
+                                int format=0,
+                                int channelCount=0,
+                                uint32_t sampleRate=0);
+
+    virtual AudioStreamIn* openInputStream(
+                                int format,
+                                int channelCount,
+                                uint32_t sampleRate);
+
+protected:
+    virtual status_t    doRouting() { return NO_ERROR; }
+    virtual status_t    dump(int fd, const Vector<String16>& args);
+
+            bool        mMicMute;
+private:
+    status_t            dumpInternals(int fd, const Vector<String16>& args);
+};
+
+// ----------------------------------------------------------------------------
+
+}; // namespace android
+
+#endif // ANDROID_AUDIO_HARDWARE_STUB_H