Merge "Don't call virtual functions in the destructor for audio and camera source classes"
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 1417416..7b0b443 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -226,8 +226,8 @@
AudioEffect(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -238,8 +238,8 @@
AudioEffect(const char *typeStr,
const char *uuidStr = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -260,8 +260,8 @@
status_t set(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 76ec3b1..c8c5dba 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -155,8 +155,8 @@
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
@@ -181,8 +181,8 @@
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
int sessionId = 0);
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 6b12c14..74a1e62 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -248,7 +248,7 @@
static sp<IAudioPolicyService> gAudioPolicyService;
// mapping between stream types and outputs
- static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
+ static DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> gStreamOutputMap;
// list of output descriptors containing cached parameters
// (sampling rate, framecount, channel count...)
static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 98abfbd..02c85cd 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -148,8 +148,8 @@
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
@@ -180,8 +180,8 @@
int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
@@ -204,8 +204,8 @@
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 7c0d886..760595c 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -126,7 +126,7 @@
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics) = 0;
+ audio_in_acoustics_t acoustics) = 0;
virtual status_t closeInput(int input) = 0;
virtual status_t setStreamOutput(audio_stream_type_t stream, int output) = 0;
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 7d890bd..df0c97e 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -154,7 +154,7 @@
ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava = false);
~ToneGenerator();
- bool startTone(int toneType, int durationMs = -1);
+ bool startTone(tone_type toneType, int durationMs = -1);
void stopTone();
bool isInited() { return (mState == TONE_IDLE)?false:true;}
@@ -274,7 +274,7 @@
bool prepareWave();
unsigned int numWaves(unsigned int segmentIdx);
void clearWaveGens();
- int getToneForRegion(int toneType);
+ tone_type getToneForRegion(tone_type toneType);
// WaveGenerator generates a single sine wave
class WaveGenerator {
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index 1a4cbca..60fa15b 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -66,8 +66,8 @@
* See AudioEffect constructor for details on parameters.
*/
Visualizer(int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0);
~Visualizer();
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index fa3ab8a..49397ee 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -34,7 +34,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -66,6 +66,7 @@
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
status_t writeAdtsHeader(uint32_t frameLength);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(AACWriter);
};
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 62d57b4..392f968 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -60,6 +60,7 @@
status_t threadFunc();
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
+ status_t reset();
AMRWriter(const AMRWriter &);
AMRWriter &operator=(const AMRWriter &);
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index e4c1c49..a7c9ecf 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -78,6 +78,7 @@
void writeAccessUnit(int32_t sourceIndex, const sp<ABuffer> &buffer);
ssize_t internalWrite(const void *data, size_t size);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSWriter);
};
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 77166ed..0409b30 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -184,6 +184,7 @@
void writeLongitude(int degreex10000);
void sendSessionSummary();
void release();
+ status_t reset();
MPEG4Writer(const MPEG4Writer &);
MPEG4Writer &operator=(const MPEG4Writer &);
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ffc546e..dd97ce4 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -76,7 +76,9 @@
// Left channel is in [0:15], right channel is in [16:31].
// Always read and write the combined pair atomically.
// For AudioTrack only, not used by AudioRecord.
- uint32_t volumeLR;
+private:
+ uint32_t mVolumeLR;
+public:
uint32_t sampleRate;
// NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
@@ -116,6 +118,17 @@
uint16_t getSendLevel_U4_12() const {
return mSendLevel;
}
+
+ // for AudioTrack client only, caller must limit to 0 <= volumeLR <= 0x10001000
+ void setVolumeLR(uint32_t volumeLR) {
+ mVolumeLR = volumeLR;
+ }
+
+ // for AudioFlinger only; the return value must be validated by the caller
+ uint32_t getVolumeLR() const {
+ return mVolumeLR;
+ }
+
};
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index 77d40b6..7f7c7e1 100755
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -13,7 +13,7 @@
LOCAL_C_INCLUDES += \
external/webrtc/src \
external/webrtc/src/modules/interface \
- external/webrtc/src/modules/audio_processing/main/interface \
+ external/webrtc/src/modules/audio_processing/interface \
system/media/audio_effects/include
LOCAL_C_INCLUDES += $(call include-path-for, speex)
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index e988e06..9fd6764 100755
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -24,8 +24,8 @@
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_agc.h>
#include <audio_effects/effect_ns.h>
-#include "modules/interface/module_common_types.h"
-#include "modules/audio_processing/main/interface/audio_processing.h"
+#include <module_common_types.h>
+#include <audio_processing.h>
#include "speex/speex_resampler.h"
@@ -220,8 +220,8 @@
// Automatic Gain Control (AGC)
//------------------------------------------------------------------------------
-static const int kAgcDefaultTargetLevel = 0;
-static const int kAgcDefaultCompGain = 90;
+static const int kAgcDefaultTargetLevel = 3;
+static const int kAgcDefaultCompGain = 9;
static const bool kAgcDefaultLimiter = true;
int AgcInit (preproc_effect_t *effect)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 6639d06..a242846 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -342,7 +342,7 @@
{
ALOGW("IEffect died");
mStatus = NO_INIT;
- if (mCbf) {
+ if (mCbf != NULL) {
status_t status = DEAD_OBJECT;
mCbf(EVENT_ERROR, mUserData, &status);
}
@@ -363,7 +363,7 @@
mStatus = ALREADY_EXISTS;
}
}
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_CONTROL_STATUS_CHANGED, mUserData, &controlGranted);
}
}
@@ -373,7 +373,7 @@
ALOGV("enableStatusChanged %p enabled %d mCbf %p", this, enabled, mCbf);
if (mStatus == ALREADY_EXISTS) {
mEnabled = enabled;
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_ENABLE_STATUS_CHANGED, mUserData, &enabled);
}
}
@@ -389,7 +389,7 @@
return;
}
- if (mCbf && cmdCode == EFFECT_CMD_SET_PARAM) {
+ if (mCbf != NULL && cmdCode == EFFECT_CMD_SET_PARAM) {
effect_param_t *cmd = (effect_param_t *)cmdData;
cmd->status = *(int32_t *)replyData;
mCbf(EVENT_PARAMETER_CHANGED, mUserData, cmd);
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 2b3ea38..c96bc76 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -206,7 +206,7 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
}
@@ -387,7 +387,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -397,7 +397,7 @@
status_t AudioRecord::getMarkerPosition(uint32_t *marker)
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -406,7 +406,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -418,7 +418,7 @@
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -427,7 +427,7 @@
status_t AudioRecord::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mCblk->user;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 124032b..110a294 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -35,7 +35,8 @@
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
// Cached values
-DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
+
+DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
// Cached values for recording queries, all protected by gLock
@@ -224,7 +225,7 @@
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -262,7 +263,7 @@
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -293,7 +294,7 @@
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -404,7 +405,7 @@
void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
ALOGV("ioConfigChanged() event %d", event);
OutputDescriptor *desc;
- uint32_t stream;
+ audio_stream_type_t stream;
if (ioHandle == 0) return;
@@ -412,8 +413,8 @@
switch (event) {
case STREAM_CONFIG_CHANGED:
- if (param2 == 0) break;
- stream = *(uint32_t *)param2;
+ if (param2 == NULL) break;
+ stream = *(audio_stream_type_t *)param2;
ALOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
if (gStreamOutputMap.indexOfKey(stream) >= 0) {
gStreamOutputMap.replaceValueFor(stream, ioHandle);
@@ -424,7 +425,7 @@
ALOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
@@ -453,7 +454,7 @@
ALOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 17e3d4b..8c33f41 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -257,7 +257,7 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
}
@@ -501,7 +501,7 @@
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
- mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
@@ -604,13 +604,13 @@
status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
{
AutoMutex lock(mLock);
- if (loopStart != 0) {
+ if (loopStart != NULL) {
*loopStart = mCblk->loopStart;
}
- if (loopEnd != 0) {
+ if (loopEnd != NULL) {
*loopEnd = mCblk->loopEnd;
}
- if (loopCount != 0) {
+ if (loopCount != NULL) {
if (mCblk->loopCount < 0) {
*loopCount = -1;
} else {
@@ -623,7 +623,7 @@
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -633,7 +633,7 @@
status_t AudioTrack::getMarkerPosition(uint32_t *marker)
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -642,7 +642,7 @@
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -654,7 +654,7 @@
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -679,7 +679,7 @@
status_t AudioTrack::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
@@ -837,7 +837,7 @@
mCblk->stepUser(mCblk->frameCount);
}
- mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
@@ -1319,8 +1319,8 @@
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
- userBase(0), serverBase(0), buffers(0), frameCount(0),
- loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
+ userBase(0), serverBase(0), buffers(NULL), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
mSendLevel(0), flags(0)
{
}
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 0d442ef..fc5520f 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -432,7 +432,7 @@
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
@@ -445,7 +445,7 @@
data.writeInt32(samplingRate);
data.writeInt32(format);
data.writeInt32(channels);
- data.writeInt32(acoustics);
+ data.writeInt32((int32_t) acoustics);
remote()->transact(OPEN_INPUT, data, &reply);
int input = reply.readInt32();
devices = reply.readInt32();
@@ -640,7 +640,7 @@
*id = tmp;
}
tmp = reply.readInt32();
- if (enabled) {
+ if (enabled != NULL) {
*enabled = tmp;
}
effect = interface_cast<IEffect>(reply.readStrongBinder());
@@ -881,13 +881,13 @@
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
- uint32_t acoutics = data.readInt32();
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t) data.readInt32();
int input = openInput(&devices,
&samplingRate,
&format,
&channels,
- acoutics);
+ acoustics);
reply->writeInt32(input);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 5a3f250..9458bc0 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -73,7 +73,7 @@
CHECK_INTERFACE(IAudioFlingerClient, data, reply);
int event = data.readInt32();
int ioHandle = data.readInt32();
- void *param2 = 0;
+ void *param2 = NULL;
AudioSystem::OutputDescriptor desc;
uint32_t stream;
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 5ceb912..e6e989d 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -751,7 +751,7 @@
// Used by ToneGenerator::getToneForRegion() to convert user specified supervisory tone type
// to actual tone for current region.
-const unsigned char ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
+const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
{ // ANSI
TONE_ANSI_DIAL, // TONE_SUP_DIAL
TONE_ANSI_BUSY, // TONE_SUP_BUSY
@@ -811,9 +811,9 @@
mThreadCanCallJava = threadCanCallJava;
mStreamType = streamType;
mVolume = volume;
- mpAudioTrack = 0;
- mpToneDesc = 0;
- mpNewToneDesc = 0;
+ mpAudioTrack = NULL;
+ mpToneDesc = NULL;
+ mpNewToneDesc = NULL;
// Generate tone by chunks of 20 ms to keep cadencing precision
mProcessSize = (mSamplingRate * 20) / 1000;
@@ -855,7 +855,7 @@
ToneGenerator::~ToneGenerator() {
ALOGV("ToneGenerator destructor\n");
- if (mpAudioTrack) {
+ if (mpAudioTrack != NULL) {
stopTone();
ALOGV("Delete Track: %p\n", mpAudioTrack);
delete mpAudioTrack;
@@ -878,7 +878,7 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-bool ToneGenerator::startTone(int toneType, int durationMs) {
+bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
bool lResult = false;
status_t lStatus;
@@ -1012,7 +1012,7 @@
if (mpAudioTrack) {
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
@@ -1048,7 +1048,7 @@
if (mpAudioTrack) {
ALOGV("Delete Track I: %p\n", mpAudioTrack);
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
return false;
@@ -1317,7 +1317,7 @@
bool ToneGenerator::prepareWave() {
unsigned int segmentIdx = 0;
- if (!mpNewToneDesc) {
+ if (mpNewToneDesc == NULL) {
return false;
}
@@ -1434,13 +1434,13 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-int ToneGenerator::getToneForRegion(int toneType) {
- int regionTone;
+ToneGenerator::tone_type ToneGenerator::getToneForRegion(tone_type toneType) {
+ tone_type regionTone;
if (mRegion == CEPT || toneType < FIRST_SUP_TONE || toneType > LAST_SUP_TONE) {
regionTone = toneType;
} else {
- regionTone = sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
+ regionTone = (tone_type) sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
}
ALOGV("getToneForRegion, tone %d, region %d, regionTone %d", toneType, mRegion, regionTone);
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 1673ccd..9cdb463 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -60,7 +60,7 @@
AACWriter::~AACWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AACWriter::stop() {
+status_t AACWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 6c4e307..59b4ca7 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -52,7 +52,7 @@
AMRWriter::~AMRWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AMRWriter::stop() {
+status_t AMRWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index a452ad5..483e5ab 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -78,8 +78,7 @@
LOCAL_STATIC_LIBRARIES := \
libstagefright_color_conversion \
- libstagefright_amrnbenc \
- libstagefright_amrwbenc \
+ libstagefright_aacenc \
libstagefright_avcenc \
libstagefright_m4vh263enc \
libstagefright_matroska \
@@ -141,7 +140,6 @@
################################################################################
LOCAL_SHARED_LIBRARIES += \
- libstagefright_amrnb_common \
libstagefright_enc_common \
libstagefright_avc_common \
libstagefright_foundation \
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 36009ab..0b4ecbe 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -513,7 +513,7 @@
MPEG2TSWriter::~MPEG2TSWriter() {
if (mStarted) {
- stop();
+ reset();
}
mLooper->unregisterHandler(mReflector->id());
@@ -564,7 +564,7 @@
return OK;
}
-status_t MPEG2TSWriter::stop() {
+status_t MPEG2TSWriter::reset() {
CHECK(mStarted);
for (size_t i = 0; i < mSources.size(); ++i) {
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 06dd875..068660b 100755
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -282,7 +282,7 @@
}
MPEG4Writer::~MPEG4Writer() {
- stop();
+ reset();
while (!mTracks.empty()) {
List<Track *>::iterator it = mTracks.begin();
@@ -616,7 +616,7 @@
mStarted = false;
}
-status_t MPEG4Writer::stop() {
+status_t MPEG4Writer::reset() {
if (mInitCheck != OK) {
return OK;
} else {
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 1f26cbe..381320b 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -18,8 +18,7 @@
#define LOG_TAG "OMXCodec"
#include <utils/Log.h>
-#include "include/AMRNBEncoder.h"
-#include "include/AMRWBEncoder.h"
+#include "include/AACEncoder.h"
#include "include/AVCEncoder.h"
#include "include/M4vH263Encoder.h"
@@ -70,8 +69,7 @@
#define FACTORY_REF(name) { #name, Make##name },
-FACTORY_CREATE_ENCODER(AMRNBEncoder)
-FACTORY_CREATE_ENCODER(AMRWBEncoder)
+FACTORY_CREATE_ENCODER(AACEncoder)
FACTORY_CREATE_ENCODER(AVCEncoder)
FACTORY_CREATE_ENCODER(M4vH263Encoder)
@@ -84,8 +82,7 @@
};
static const FactoryInfo kFactoryInfo[] = {
- FACTORY_REF(AMRNBEncoder)
- FACTORY_REF(AMRWBEncoder)
+ FACTORY_REF(AACEncoder)
FACTORY_REF(AVCEncoder)
FACTORY_REF(M4vH263Encoder)
};
@@ -146,11 +143,12 @@
static const CodecInfo kEncoderInfo[] = {
{ MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.TI.AMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_NB, "AMRNBEncoder" },
+ { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.google.amrnb.encoder" },
{ MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "AMRWBEncoder" },
+ { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.google.amrwb.encoder" },
{ MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.encode" },
{ MEDIA_MIMETYPE_AUDIO_AAC, "OMX.google.aac.encoder" },
+ { MEDIA_MIMETYPE_AUDIO_AAC, "AACEncoder" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.video.encoder.mpeg4" },
diff --git a/media/libstagefright/codecs/amrnb/enc/Android.mk b/media/libstagefright/codecs/amrnb/enc/Android.mk
index b6aed81..94e8726 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.mk
+++ b/media/libstagefright/codecs/amrnb/enc/Android.mk
@@ -74,3 +74,30 @@
LOCAL_MODULE := libstagefright_amrnbenc
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRNBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ $(LOCAL_PATH)/src \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../common/include \
+ $(LOCAL_PATH)/../common
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrnbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_amrnb_common
+
+LOCAL_MODULE := libstagefright_soft_amrnbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
new file mode 100644
index 0000000..07f8b4f
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
@@ -0,0 +1,404 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRNBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRNBEncoder.h"
+
+#include "gsmamr_enc.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 8000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRNBEncoder::SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncState(NULL),
+ mSidState(NULL),
+ mBitRate(0),
+ mMode(MR475),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRNBEncoder::~SoftAMRNBEncoder() {
+ if (mEncState != NULL) {
+ AMREncodeExit(&mEncState, &mSidState);
+ mEncState = mSidState = NULL;
+ }
+}
+
+void SoftAMRNBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/3gpp");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRNBEncoder::initEncoder() {
+ if (AMREncodeInit(&mEncState, &mSidState, false /* dtx_enable */) != 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+ amrParams->eAMRBandMode = (OMX_AUDIO_AMRBANDMODETYPE)(mMode + 1);
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrnb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeNB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeNB7) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+ mMode = amrParams->eAMRBandMode - 1;
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRNBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ Frame_Type_3GPP frameType;
+ int res = AMREncode(
+ mEncState, mSidState, (Mode)mMode,
+ mInputFrame, outPtr, &frameType, AMR_TX_WMF);
+
+ CHECK_GE(res, 0);
+ CHECK_LE((size_t)res, outAvailable);
+
+ // Convert header byte from WMF to IETF format.
+ outPtr[0] = ((outPtr[0] << 3) | 4) & 0x7c;
+
+ outHeader->nFilledLen = res;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRNBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
new file mode 100644
index 0000000..50178c4
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2012 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 SOFT_AMRNB_ENCODER_H_
+
+#define SOFT_AMRNB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+namespace android {
+
+struct SoftAMRNBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRNBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 160,
+ };
+
+ void *mEncState;
+ void *mSidState;
+
+ OMX_U32 mBitRate;
+ int mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRNBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRNB_ENCODER_H_
diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index ae43870..6ce6171 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -117,4 +117,26 @@
include $(BUILD_STATIC_LIBRARY)
+################################################################################
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRWBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media/libstagefright/codecs/common/include \
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrwbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_amrwbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
new file mode 100644
index 0000000..9ccb49c
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
@@ -0,0 +1,459 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRWBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRWBEncoder.h"
+
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 16000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRWBEncoder::SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mBitRate(0),
+ mMode(VOAMRWB_MD66),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRWBEncoder::~SoftAMRWBEncoder() {
+ if (mEncoderHandle != NULL) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAMRWBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/amr-wb");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRWBEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAMRWBEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+
+ if (VO_ERR_NONE != mApiHandle->Init(
+ &mEncoderHandle, VO_AUDIO_CodingAMRWB, &userData)) {
+ ALOGE("Failed to init AMRWB encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ VOAMRWBFRAMETYPE type = VOAMRWB_RFC3267;
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &type)) {
+ ALOGE("Failed to set AMRWB encoder frame type to %d", type);
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+
+ amrParams->eAMRBandMode =
+ (OMX_AUDIO_AMRBANDMODETYPE)(mMode + OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrwb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeWB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeWB8) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+
+ mMode = (VOAMRWBMODE)(
+ amrParams->eAMRBandMode - OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ if (VO_ERR_NONE !=
+ mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_MODE, &mMode)) {
+ ALOGE("Failed to set AMRWB encoder mode to %d", mMode);
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != (OMX_U32)kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRWBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *) mInputFrame;
+ inputData.Length = mInputSize;
+
+ CHECK_EQ(VO_ERR_NONE,
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable;
+ VO_U32 ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ CHECK(ret == VO_ERR_NONE || ret == VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = outputData.Length;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %ld bytes of data (time = %lld us, flags = 0x%08lx)",
+ outHeader->nFilledLen, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRWBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
new file mode 100644
index 0000000..d0c1dab
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2012 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 SOFT_AMRWB_ENCODER_H_
+
+#define SOFT_AMRWB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+#include "voAMRWB.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAMRWBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRWBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 320,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mBitRate;
+ VOAMRWBMODE mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRWBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRWB_ENCODER_H_
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index 5cc3f78..f3ef3de 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -144,8 +144,8 @@
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_ptr = (const uint8_t *)src.mBits
+ (src.mCropTop * dst.mWidth + src.mCropLeft) * 2;
@@ -182,11 +182,15 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_ptr += src.mWidth * 2;
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -290,15 +294,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -340,7 +343,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -349,7 +356,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -361,15 +368,14 @@
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -411,7 +417,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -420,7 +430,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -430,15 +440,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y = (const uint8_t *)src.mBits;
@@ -478,7 +487,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -487,7 +500,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
diff --git a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
index 0914f32..c79e01f 100644
--- a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
@@ -333,8 +333,9 @@
void SimpleSoftOMXComponent::onMessageReceived(const sp<AMessage> &msg) {
Mutex::Autolock autoLock(mLock);
-
- switch (msg->what()) {
+ uint32_t msgType = msg->what();
+ ALOGV("msgType = %d", msgType);
+ switch (msgType) {
case kWhatSendCommand:
{
int32_t cmd, param;
@@ -354,27 +355,27 @@
CHECK(mState == OMX_StateExecuting && mTargetState == mState);
bool found = false;
- for (size_t i = 0; i < mPorts.size(); ++i) {
- PortInfo *port = &mPorts.editItemAt(i);
+ size_t portIndex = (kWhatEmptyThisBuffer == msgType)?
+ header->nInputPortIndex: header->nOutputPortIndex;
+ PortInfo *port = &mPorts.editItemAt(portIndex);
- for (size_t j = 0; j < port->mBuffers.size(); ++j) {
- BufferInfo *buffer = &port->mBuffers.editItemAt(j);
+ for (size_t j = 0; j < port->mBuffers.size(); ++j) {
+ BufferInfo *buffer = &port->mBuffers.editItemAt(j);
- if (buffer->mHeader == header) {
- CHECK(!buffer->mOwnedByUs);
+ if (buffer->mHeader == header) {
+ CHECK(!buffer->mOwnedByUs);
- buffer->mOwnedByUs = true;
+ buffer->mOwnedByUs = true;
- CHECK((msg->what() == kWhatEmptyThisBuffer
- && port->mDef.eDir == OMX_DirInput)
- || (port->mDef.eDir == OMX_DirOutput));
+ CHECK((msgType == kWhatEmptyThisBuffer
+ && port->mDef.eDir == OMX_DirInput)
+ || (port->mDef.eDir == OMX_DirOutput));
- port->mQueue.push_back(buffer);
- onQueueFilled(i);
+ port->mQueue.push_back(buffer);
+ onQueueFilled(portIndex);
- found = true;
- break;
- }
+ found = true;
+ break;
}
}
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index cf9e8c9..99ffe7d 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -37,7 +37,9 @@
{ "OMX.google.aac.decoder", "aacdec", "audio_decoder.aac" },
{ "OMX.google.aac.encoder", "aacenc", "audio_encoder.aac" },
{ "OMX.google.amrnb.decoder", "amrdec", "audio_decoder.amrnb" },
+ { "OMX.google.amrnb.encoder", "amrnbenc", "audio_encoder.amrnb" },
{ "OMX.google.amrwb.decoder", "amrdec", "audio_decoder.amrwb" },
+ { "OMX.google.amrwb.encoder", "amrwbenc", "audio_encoder.amrwb" },
{ "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },
{ "OMX.google.g711.alaw.decoder", "g711dec", "audio_decoder.g711alaw" },
{ "OMX.google.g711.mlaw.decoder", "g711dec", "audio_decoder.g711mlaw" },
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2d856ad..f71ba0a 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -160,7 +160,10 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mPrimaryHardwareDev(NULL),
+ mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
+ mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false)
{
}
@@ -172,7 +175,6 @@
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
- mHardwareStatus = AUDIO_HW_IDLE;
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
@@ -265,13 +267,10 @@
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
- wp<Client> wClient = mClients.valueAt(i);
- if (wClient != 0) {
- sp<Client> client = wClient.promote();
- if (client != 0) {
- snprintf(buffer, SIZE, " pid: %d\n", client->pid());
- result.append(buffer);
- }
+ sp<Client> client = mClients.valueAt(i).promote();
+ if (client != 0) {
+ snprintf(buffer, SIZE, " pid: %d\n", client->pid());
+ result.append(buffer);
}
}
@@ -971,7 +970,8 @@
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
- mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
+ mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
+ param2);
}
}
@@ -985,13 +985,19 @@
// ----------------------------------------------------------------------------
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device,
+ type_t type)
: Thread(false),
- mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
- mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false),
- mDevice(device)
+ mType(type),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
+ // mChannelMask
+ mChannelCount(0),
+ mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mParamStatus(NO_ERROR),
+ mStandby(false), mId(id), mExiting(false),
+ mDevice(device),
+ mDeathRecipient(new PMDeathRecipient(this))
{
- mDeathRecipient = new PMDeathRecipient(this);
}
AudioFlinger::ThreadBase::~ThreadBase()
@@ -1372,20 +1378,24 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
int id,
- uint32_t device)
- : ThreadBase(audioFlinger, id, device),
- mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
+ uint32_t device,
+ type_t type)
+ : ThreadBase(audioFlinger, id, device, type),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterMute as parameter
+ mMasterMute(audioFlinger->masterMute_l()),
+ // mStreamTypes[] initialized in constructor body
+ mOutput(output),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterVolume as parameter
+ mMasterVolume(audioFlinger->masterVolume_l()),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
readOutputParameters();
- // Assumes constructor is called by AudioFlinger with it's mLock held,
- // but it would be safer to explicitly pass these as parameters
- mMasterVolume = mAudioFlinger->masterVolume_l();
- mMasterMute = mAudioFlinger->masterMute_l();
-
// mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
// There is no AUDIO_STREAM_MIN, and ++ operator does not compile
for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
@@ -1431,13 +1441,10 @@
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
- wp<Track> wTrack = mActiveTracks[i];
- if (wTrack != 0) {
- sp<Track> track = wTrack.promote();
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
}
}
write(fd, result.string(), result.size());
@@ -1705,7 +1712,7 @@
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
@@ -1740,7 +1747,7 @@
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
- if (mMixBuffer != NULL) delete[] mMixBuffer;
+ delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
@@ -1758,7 +1765,7 @@
status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
{
- if (halFrames == 0 || dspFrames == 0) {
+ if (halFrames == NULL || dspFrames == NULL) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
@@ -1845,13 +1852,12 @@
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE)
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ int id, uint32_t device, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type),
+ mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
+ mPrevMixerStatus(MIXER_IDLE)
{
- mType = ThreadBase::MIXER;
- mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
-
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
ALOGE("Invalid audio hardware channel count");
@@ -2193,7 +2199,7 @@
// read original volumes with volume control
float typeVolume = mStreamTypes[track->type()].volume;
float v = masterVolume * typeVolume;
- uint32_t vlr = cblk->volumeLR;
+ uint32_t vlr = cblk->getVolumeLR();
vl = vlr & 0xFFFF;
vr = vlr >> 16;
// track volumes come from shared memory, so can't be trusted and must be clamped
@@ -2461,6 +2467,8 @@
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
+ // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+ mAudioMixer = NULL;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -2513,9 +2521,10 @@
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ // mLeftVolFloat, mRightVolFloat
+ // mLeftVolShort, mRightVolShort
{
- mType = ThreadBase::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
@@ -2729,7 +2738,7 @@
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
- uint32_t vlr = cblk->volumeLR;
+ uint32_t vlr = cblk->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
@@ -2992,10 +3001,11 @@
// ----------------------------------------------------------------------------
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ AudioFlinger::MixerThread* mainThread, int id)
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
+ mWaitTimeMs(UINT_MAX)
{
- mType = ThreadBase::DUPLICATING;
addOutputTrack(mainThread);
}
@@ -3244,13 +3254,17 @@
: RefBase(),
mThread(thread),
mClient(client),
- mCblk(0),
+ mCblk(NULL),
+ // mBuffer
+ // mBufferEnd
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
+ // mChannelCount
+ // mChannelMask
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
@@ -3266,7 +3280,7 @@
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) { // construct the shared structure in-place.
+ if (mCblk != NULL) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
@@ -3309,7 +3323,7 @@
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
- if (mCblk) {
+ if (mCblk != NULL) {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
if (mClient == NULL) {
delete mCblk;
@@ -3317,6 +3331,7 @@
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != NULL) {
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
@@ -3383,7 +3398,7 @@
server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
- return 0;
+ return NULL;
}
return bufferStart;
@@ -3468,7 +3483,7 @@
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
- uint32_t vlr = mCblk->volumeLR;
+ uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
@@ -3827,7 +3842,6 @@
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
@@ -4065,7 +4079,7 @@
mAudioFlinger->removeClient_l(mPid);
}
-const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
@@ -4075,13 +4089,12 @@
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
- : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
+ : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
- mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
@@ -4266,15 +4279,16 @@
uint32_t channels,
int id,
uint32_t device) :
- ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
+ ThreadBase(audioFlinger, id, device, RECORD),
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInIndex and mInputBytes set by readInputParameters()
+ mReqChannelCount(popcount(channels)),
+ mReqSampleRate(sampleRate)
+ // mBytesRead is only meaningful while active, and so is cleared in start()
+ // (but might be better to also clear here for dump?)
{
- mType = ThreadBase::RECORD;
-
snprintf(mName, kNameLength, "AudioIn_%d", id);
- mReqChannelCount = popcount(channels);
- mReqSampleRate = sampleRate;
readInputParameters();
}
@@ -4282,10 +4296,8 @@
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != NULL) {
- delete mResampler;
- delete[] mRsmpOutBuffer;
- }
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
}
void AudioFlinger::RecordThread::onFirstRef()
@@ -4807,7 +4819,7 @@
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
switch (event) {
case AudioSystem::INPUT_OPENED:
@@ -4829,9 +4841,11 @@
void AudioFlinger::RecordThread::readInputParameters()
{
- if (mRsmpInBuffer) delete mRsmpInBuffer;
- if (mRsmpOutBuffer) delete mRsmpOutBuffer;
- if (mResampler) delete mResampler;
+ delete mRsmpInBuffer;
+ // mRsmpInBuffer is always assigned a new[] below
+ delete mRsmpOutBuffer;
+ mRsmpOutBuffer = NULL;
+ delete mResampler;
mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
@@ -4983,10 +4997,10 @@
}
mPlaybackThreads.add(id, thread);
- if (pSamplingRate) *pSamplingRate = samplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = channels;
- if (pLatencyMs) *pLatencyMs = thread->latency();
+ if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = channels;
+ if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
@@ -5038,7 +5052,7 @@
}
}
}
- void *param2 = 0;
+ void *param2 = NULL;
audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
mPlaybackThreads.removeItem(output);
}
@@ -5089,7 +5103,7 @@
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
@@ -5114,7 +5128,7 @@
status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
inStream,
@@ -5134,7 +5148,7 @@
ALOGV("openInput() reopening with proposed sampling rate and channels");
status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
}
@@ -5154,9 +5168,9 @@
device);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
- if (pSamplingRate) *pSamplingRate = reqSamplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = reqChannels;
+ if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = reqChannels;
input->stream->common.standby(&input->stream->common);
@@ -5181,7 +5195,7 @@
}
ALOGV("closeInput() %d", input);
- void *param2 = 0;
+ void *param2 = NULL;
audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
mRecordThreads.removeItem(input);
}
@@ -5243,12 +5257,8 @@
return;
}
}
- AudioSessionRef *ref = new AudioSessionRef();
- ref->sessionid = audioSession;
- ref->pid = caller;
- ref->cnt = 1;
- mAudioSessionRefs.push(ref);
- ALOGV(" added new entry for %d", ref->sessionid);
+ mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
+ ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(int audioSession)
@@ -5792,7 +5802,7 @@
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
- if (enabled) {
+ if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
}
}
@@ -6179,7 +6189,7 @@
}
}
-status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
+status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
{
status_t status;
@@ -6226,7 +6236,7 @@
bool enabled = false;
EffectHandle *hdl = handle.unsafe_get();
- if (hdl) {
+ if (hdl != NULL) {
ALOGV("removeHandle() unsafe_get OK");
enabled = hdl->enabled();
}
@@ -6862,7 +6872,7 @@
if (mCblkMemory != 0) {
mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) {
+ if (mCblk != NULL) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
}
@@ -6959,7 +6969,7 @@
// release sp on module => module destructor can be called now
mEffect.clear();
if (mClient != 0) {
- if (mCblk) {
+ if (mCblk != NULL) {
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
mCblkMemory.clear(); // and free the shared memory
@@ -7089,7 +7099,7 @@
void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
{
- bool locked = mCblk ? tryLock(mCblk->lock) : false;
+ bool locked = mCblk != NULL && tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
(mClient == NULL) ? getpid() : mClient->pid(),
@@ -7551,7 +7561,8 @@
ALOGV("setEffectSuspendedAll_l() add entry for 0");
}
if (desc->mRefCount++ == 0) {
- Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
+ Vector< sp<EffectModule> > effects;
+ getSuspendEligibleEffects(effects);
for (size_t i = 0; i < effects.size(); i++) {
setEffectSuspended_l(&effects[i]->desc().type, true);
}
@@ -7602,16 +7613,14 @@
return true;
}
-Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
{
- Vector< sp<EffectModule> > effects;
+ effects.clear();
for (size_t i = 0; i < mEffects.size(); i++) {
- if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
- continue;
+ if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
+ effects.add(mEffects[i]);
}
- effects.add(mEffects[i]);
}
- return effects;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 766ba44..3f3188c 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -131,7 +131,7 @@
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics);
+ audio_in_acoustics_t acoustics);
virtual status_t closeInput(int input);
@@ -226,16 +226,16 @@
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
- const sp<MemoryDealer>& heap() const;
+ sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
private:
Client(const Client&);
Client& operator = (const Client&);
- sp<AudioFlinger> mAudioFlinger;
- sp<MemoryDealer> mMemoryDealer;
- pid_t mPid;
+ const sp<AudioFlinger> mAudioFlinger;
+ const sp<MemoryDealer> mMemoryDealer;
+ const pid_t mPid;
};
// --- Notification Client ---
@@ -246,7 +246,7 @@
pid_t pid);
virtual ~NotificationClient();
- sp<IAudioFlingerClient> client() { return mClient; }
+ sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
@@ -255,9 +255,9 @@
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
- sp<AudioFlinger> mAudioFlinger;
- pid_t mPid;
- sp<IAudioFlingerClient> mClient;
+ const sp<AudioFlinger> mAudioFlinger;
+ const pid_t mPid;
+ const sp<IAudioFlingerClient> mAudioFlingerClient;
};
class TrackHandle;
@@ -277,17 +277,17 @@
class ThreadBase : public Thread {
public:
- ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device);
- virtual ~ThreadBase();
-
- enum type {
+ enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD // Thread class is RecordThread
};
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, type_t type);
+ virtual ~ThreadBase();
+
status_t dumpBase(int fd, const Vector<String16>& args);
status_t dumpEffectChains(int fd, const Vector<String16>& args);
@@ -367,8 +367,8 @@
bool step();
void reset();
- wp<ThreadBase> mThread;
- sp<Client> mClient;
+ const wp<ThreadBase> mThread;
+ /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
@@ -377,9 +377,9 @@
// we don't really need a lock for these
track_state mState;
int mClientTid;
- audio_format_t mFormat;
+ const audio_format_t mFormat;
uint32_t mFlags;
- int mSessionId;
+ const int mSessionId;
uint8_t mChannelCount;
uint32_t mChannelMask;
};
@@ -408,7 +408,7 @@
};
virtual status_t initCheck() const = 0;
- int type() const { return mType; }
+ type_t type() const { return mType; }
uint32_t sampleRate() const;
int channelCount() const;
audio_format_t format() const;
@@ -530,9 +530,9 @@
friend class RecordThread;
friend class RecordTrack;
- int mType;
+ const type_t mType;
Condition mWaitWorkCV;
- sp<AudioFlinger> mAudioFlinger;
+ const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannelMask;
@@ -553,7 +553,7 @@
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
- sp<PMDeathRecipient> mDeathRecipient;
+ const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
@@ -671,7 +671,7 @@
bool write(int16_t* data, uint32_t frames);
bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
bool isActive() { return mActive; }
- wp<ThreadBase>& thread() { return mThread; }
+ const wp<ThreadBase>& thread() { return mThread; }
private:
@@ -688,10 +688,11 @@
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
- DuplicatingThread* mSourceThread;
+ DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
- PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id,
+ uint32_t device, type_t type);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -817,7 +818,8 @@
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
int id,
- uint32_t device);
+ uint32_t device,
+ type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
@@ -917,7 +919,7 @@
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<PlaybackThread::Track> mTrack;
+ const sp<PlaybackThread::Track> mTrack;
};
friend class Client;
@@ -1021,8 +1023,8 @@
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
- int mReqChannelCount;
- uint32_t mReqSampleRate;
+ const int mReqChannelCount;
+ const uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
@@ -1036,7 +1038,7 @@
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<RecordThread::RecordTrack> mRecordTrack;
+ const sp<RecordThread::RecordTrack> mRecordTrack;
};
//--- Audio Effect Management
@@ -1105,9 +1107,9 @@
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
- wp<ThreadBase>& thread() { return mThread; }
+ const wp<ThreadBase>& thread() { return mThread; }
- status_t addHandle(sp<EffectHandle>& handle);
+ status_t addHandle(const sp<EffectHandle>& handle);
void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
size_t removeHandle (const wp<EffectHandle>& handle);
@@ -1325,7 +1327,8 @@
// get a list of effect modules to suspend when an effect of the type
// passed is enabled.
- Vector< sp<EffectModule> > getSuspendEligibleEffects();
+ void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
+
// get an effect module if it is currently enable
sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
// true if the effect whose descriptor is passed can be suspended
@@ -1377,8 +1380,11 @@
};
struct AudioSessionRef {
- int sessionid;
- pid_t pid;
+ // FIXME rename parameter names when fields get "m" prefix
+ AudioSessionRef(int sessionid_, pid_t pid_) :
+ sessionid(sessionid_), pid(pid_), cnt(1) {}
+ const int sessionid;
+ const pid_t pid;
int cnt;
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index a8102e5..0b9f8ba 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -48,9 +48,10 @@
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
+ mState.hook = process__nop;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
- mState.hook = process__nop;
+ // mState.reserved
track_t* t = mState.tracks;
for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
@@ -70,12 +71,13 @@
t->enabled = 0;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
- t->buffer.raw = 0;
t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // t->buffer.frameCount
t->hook = NULL;
+ t->in = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = NULL;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
@@ -123,7 +125,7 @@
track.enabled = 0;
invalidateState(1<<name);
}
- if (track.resampler) {
+ if (track.resampler != NULL) {
// delete the resampler
delete track.resampler;
track.resampler = NULL;
@@ -807,7 +809,7 @@
while (outFrames) {
t1.buffer.frameCount = outFrames;
t1.bufferProvider->getNextBuffer(&t1.buffer);
- if (!t1.buffer.raw) break;
+ if (t1.buffer.raw == NULL) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
}
@@ -1127,9 +1129,7 @@
}
}
- if (buff != NULL) {
- delete [] buff;
- }
+ delete [] buff;
}
#endif
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 2df1385..1dddbb3 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -144,9 +144,9 @@
}
mInputs.clear();
- if (mpAudioPolicy && mpAudioPolicyDev)
+ if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL)
mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
- if (mpAudioPolicyDev)
+ if (mpAudioPolicyDev != NULL)
audio_policy_dev_close(mpAudioPolicyDev);
}
@@ -649,7 +649,7 @@
release_wake_lock(mName.string());
}
mAudioCommands.clear();
- if (mpToneGenerator != NULL) delete mpToneGenerator;
+ delete mpToneGenerator;
}
void AudioPolicyService::AudioCommandThread::onFirstRef()
@@ -682,8 +682,7 @@
ToneData *data = (ToneData *)command->mParam;
ALOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
- if (mpToneGenerator != NULL)
- delete mpToneGenerator;
+ delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
@@ -790,7 +789,8 @@
return NO_ERROR;
}
-void AudioPolicyService::AudioCommandThread::startToneCommand(int type, audio_stream_type_t stream)
+void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream)
{
AudioCommand *command = new AudioCommand();
command->mCommand = START_TONE;
@@ -1160,7 +1160,7 @@
if (param == NULL && value == NULL) {
// try to parse simple parameter form {int int}
param = root->first_child;
- if (param) {
+ if (param != NULL) {
// Note: that a pair of random strings is read as 0 0
int *ptr = (int *)fx_param->data;
int *ptr2 = (int *)((char *)param + sizeof(effect_param_t));
@@ -1419,7 +1419,7 @@
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == NULL) {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 3c0f5ed..62219e5 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -79,7 +79,7 @@
audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_in_acoustics_t acoustics =
- (audio_in_acoustics_t)0,
+ (audio_in_acoustics_t)0 /*AUDIO_IN_ACOUSTICS_NONE*/,
int audioSession = 0);
virtual status_t startInput(audio_io_handle_t input);
virtual status_t stopInput(audio_io_handle_t input);
@@ -171,7 +171,8 @@
virtual bool threadLoop();
void exit();
- void startToneCommand(int type = 0, audio_stream_type_t stream = AUDIO_STREAM_VOICE_CALL);
+ void startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream);
void stopToneCommand();
status_t volumeCommand(audio_stream_type_t stream, float volume, int output, int delayMs = 0);
status_t parametersCommand(int ioHandle, const char *keyValuePairs, int delayMs = 0);
@@ -198,7 +199,7 @@
class ToneData {
public:
- int mType; // tone type (START_TONE only)
+ ToneGenerator::tone_type mType; // tone type (START_TONE only)
audio_stream_type_t mStream; // stream type (START_TONE only)
};