Merge change I93f500a5 into eclair
* changes:
Fix issue 2203561: Sholes: audio playing out of earpiece.
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 503cb31..008468c 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -314,6 +314,11 @@
};
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
+ status_t openRecord(uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags);
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
@@ -341,6 +346,7 @@
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
audio_io_handle_t mInput;
+ uint32_t mFlags;
};
}; // namespace android
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 981c2f6..14b30ae 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -391,6 +391,14 @@
};
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+ status_t createTrack(int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ const sp<IMemory>& sharedBuffer,
+ audio_io_handle_t output);
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index c884c2c..1ad1f26 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -151,7 +151,7 @@
NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1
};
- ToneGenerator(int streamType, float volume);
+ ToneGenerator(int streamType, float volume, bool threadCanCallJava = false);
~ToneGenerator();
bool startTone(int toneType, int durationMs = -1);
@@ -242,6 +242,7 @@
static const ToneDescriptor sToneDescriptors[];
+ bool mThreadCanCallJava;
unsigned int mTotalSmp; // Total number of audio samples played (gives current time)
unsigned int mNextSegSmp; // Position of next segment transition expressed in samples
// NOTE: because mTotalSmp, mNextSegSmp are stored on 32 bit, current design will operate properly
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 5e35564..e63c0d2 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -101,11 +101,6 @@
return INVALID_OPERATION;
}
- const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
- if (audioFlinger == 0) {
- return NO_INIT;
- }
-
if (inputSource == AUDIO_SOURCE_DEFAULT) {
inputSource = AUDIO_SOURCE_MIC;
}
@@ -171,22 +166,14 @@
notificationFrames = frameCount/2;
}
- // open record channel
- status_t status;
- sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,
- sampleRate, format,
- channelCount,
- frameCount,
- ((uint16_t)flags) << 16,
- &status);
- if (record == 0) {
- LOGE("AudioFlinger could not create record track, status: %d", status);
+ // create the IAudioRecord
+ status_t status = openRecord(sampleRate, format, channelCount,
+ frameCount, flags);
+
+ if (status != NO_ERROR) {
return status;
}
- sp<IMemory> cblk = record->getCblk();
- if (cblk == 0) {
- return NO_INIT;
- }
+
if (cbf != 0) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
if (mClientRecordThread == 0) {
@@ -196,11 +183,6 @@
mStatus = NO_ERROR;
- mAudioRecord = record;
- mCblkMemory = cblk;
- mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->out = 0;
mFormat = format;
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
@@ -217,6 +199,7 @@
mNewPosition = 0;
mUpdatePeriod = 0;
mInputSource = (uint8_t)inputSource;
+ mFlags = flags;
return NO_ERROR;
}
@@ -284,15 +267,26 @@
if (android_atomic_or(1, &mActive) == 0) {
ret = AudioSystem::startInput(mInput);
if (ret == NO_ERROR) {
- mNewPosition = mCblk->user + mUpdatePeriod;
- mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
- mCblk->waitTimeMs = 0;
- if (t != 0) {
- t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
- } else {
- setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
- }
ret = mAudioRecord->start();
+ if (ret == DEAD_OBJECT) {
+ LOGV("start() dead IAudioRecord: creating a new one");
+ ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags);
+ }
+ if (ret == NO_ERROR) {
+ mNewPosition = mCblk->user + mUpdatePeriod;
+ mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
+ if (t != 0) {
+ t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ } else {
+ setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ }
+ } else {
+ LOGV("start() failed");
+ AudioSystem::stopInput(mInput);
+ android_atomic_and(~1, &mActive);
+ }
}
}
@@ -396,10 +390,48 @@
// -------------------------------------------------------------------------
+status_t AudioRecord::openRecord(
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags)
+{
+ status_t status;
+ const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+ if (audioFlinger == 0) {
+ return NO_INIT;
+ }
+
+ sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,
+ sampleRate, format,
+ channelCount,
+ frameCount,
+ ((uint16_t)flags) << 16,
+ &status);
+ if (record == 0) {
+ LOGE("AudioFlinger could not create record track, status: %d", status);
+ return status;
+ }
+ sp<IMemory> cblk = record->getCblk();
+ if (cblk == 0) {
+ LOGE("Could not get control block");
+ return NO_INIT;
+ }
+ mAudioRecord.clear();
+ mAudioRecord = record;
+ mCblkMemory.clear();
+ mCblkMemory = cblk;
+ mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ mCblk->out = 0;
+
+ return NO_ERROR;
+}
+
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
- int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -411,25 +443,40 @@
uint32_t framesReady = cblk->framesReady();
if (framesReady == 0) {
- Mutex::Autolock _l(cblk->lock);
+ cblk->lock.lock();
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
- if (UNLIKELY(!active))
+ if (UNLIKELY(!active)) {
+ cblk->lock.unlock();
return NO_MORE_BUFFERS;
- if (UNLIKELY(!waitCount))
+ }
+ if (UNLIKELY(!waitCount)) {
+ cblk->lock.unlock();
return WOULD_BLOCK;
- timeout = 0;
+ }
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
- timeout = 1;
+ cblk->lock.unlock();
+ result = mAudioRecord->start();
+ if (result == DEAD_OBJECT) {
+ LOGW("obtainBuffer() dead IAudioRecord: creating a new one");
+ result = openRecord(cblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags);
+ if (result == NO_ERROR) {
+ cblk = mCblk;
+ cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ }
+ }
+ cblk->lock.lock();
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
+ cblk->lock.unlock();
return TIMED_OUT;
}
}
@@ -437,13 +484,9 @@
start_loop_here:
framesReady = cblk->framesReady();
}
+ cblk->lock.unlock();
}
- LOGW_IF(timeout,
- "*** SERIOUS WARNING *** obtainBuffer() timed out "
- "but didn't need to be locked. We recovered, but "
- "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
-
cblk->waitTimeMs = 0;
if (framesReq > framesReady) {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 4b9d272..8529a8e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -122,11 +122,6 @@
return INVALID_OPERATION;
}
- const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
- if (audioFlinger == 0) {
- LOGE("Could not get audioflinger");
- return NO_INIT;
- }
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
@@ -217,28 +212,16 @@
}
}
- // create the track
- status_t status;
- sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
- streamType,
- sampleRate,
- format,
- channelCount,
- frameCount,
- ((uint16_t)flags) << 16,
- sharedBuffer,
- output,
- &status);
+ mVolume[LEFT] = 1.0f;
+ mVolume[RIGHT] = 1.0f;
+ // create the IAudioTrack
+ status_t status = createTrack(streamType, sampleRate, format, channelCount,
+ frameCount, flags, sharedBuffer, output);
- if (track == 0) {
- LOGE("AudioFlinger could not create track, status: %d", status);
+ if (status != NO_ERROR) {
return status;
}
- sp<IMemory> cblk = track->getCblk();
- if (cblk == 0) {
- LOGE("Could not get control block");
- return NO_INIT;
- }
+
if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
@@ -249,22 +232,6 @@
mStatus = NO_ERROR;
- mAudioTrack = track;
- mCblkMemory = cblk;
- mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->out = 1;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount;
- if (sharedBuffer == 0) {
- mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- } else {
- mCblk->buffers = sharedBuffer->pointer();
- // Force buffer full condition as data is already present in shared memory
- mCblk->stepUser(mFrameCount);
- }
- mCblk->volume[0] = mCblk->volume[1] = 0x1000;
- mVolume[LEFT] = 1.0f;
- mVolume[RIGHT] = 1.0f;
mStreamType = streamType;
mFormat = format;
mChannels = channels;
@@ -351,16 +318,27 @@
}
if (android_atomic_or(1, &mActive) == 0) {
- AudioSystem::startOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
- mNewPosition = mCblk->server + mUpdatePeriod;
- mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
- mCblk->waitTimeMs = 0;
- if (t != 0) {
- t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
- } else {
- setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ audio_io_handle_t output = AudioTrack::getOutput();
+ status_t status = mAudioTrack->start();
+ if (status == DEAD_OBJECT) {
+ LOGV("start() dead IAudioTrack: creating a new one");
+ status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags, mSharedBuffer, output);
}
- mAudioTrack->start();
+ if (status == NO_ERROR) {
+ AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
+ mNewPosition = mCblk->server + mUpdatePeriod;
+ mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
+ if (t != 0) {
+ t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ } else {
+ setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ }
+ } else {
+ LOGV("start() failed");
+ android_atomic_and(~1, &mActive);
+ }
}
if (t != 0) {
@@ -617,10 +595,67 @@
// -------------------------------------------------------------------------
+status_t AudioTrack::createTrack(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ const sp<IMemory>& sharedBuffer,
+ audio_io_handle_t output)
+{
+ status_t status;
+ const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+ if (audioFlinger == 0) {
+ LOGE("Could not get audioflinger");
+ return NO_INIT;
+ }
+
+ sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
+ streamType,
+ sampleRate,
+ format,
+ channelCount,
+ frameCount,
+ ((uint16_t)flags) << 16,
+ sharedBuffer,
+ output,
+ &status);
+
+ if (track == 0) {
+ LOGE("AudioFlinger could not create track, status: %d", status);
+ return status;
+ }
+ sp<IMemory> cblk = track->getCblk();
+ if (cblk == 0) {
+ LOGE("Could not get control block");
+ return NO_INIT;
+ }
+ mAudioTrack.clear();
+ mAudioTrack = track;
+ mCblkMemory.clear();
+ mCblkMemory = cblk;
+ mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
+ mCblk->out = 1;
+ // Update buffer size in case it has been limited by AudioFlinger during track creation
+ mFrameCount = mCblk->frameCount;
+ if (sharedBuffer == 0) {
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ } else {
+ mCblk->buffers = sharedBuffer->pointer();
+ // Force buffer full condition as data is already present in shared memory
+ mCblk->stepUser(mFrameCount);
+ }
+
+ mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
+
+ return NO_ERROR;
+}
+
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
- int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -632,17 +667,20 @@
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
- Mutex::Autolock _l(cblk->lock);
+ cblk->lock.lock();
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (UNLIKELY(!active)) {
LOGV("Not active and NO_MORE_BUFFERS");
+ cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount))
+ if (UNLIKELY(!waitCount)) {
+ cblk->lock.unlock();
return WOULD_BLOCK;
- timeout = 0;
+ }
+
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
@@ -654,14 +692,23 @@
"user=%08x, server=%08x", this, cblk->user, cblk->server);
//unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
- mAudioTrack->start();
+ result = mAudioTrack->start();
+ if (result == DEAD_OBJECT) {
+ LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
+ result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags, mSharedBuffer, getOutput());
+ if (result == NO_ERROR) {
+ cblk = mCblk;
+ cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ }
+ }
cblk->lock.lock();
- timeout = 1;
}
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
+ cblk->lock.unlock();
return TIMED_OUT;
}
}
@@ -669,6 +716,7 @@
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
+ cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
@@ -684,11 +732,6 @@
framesReq = bufferEnd - u;
}
- LOGW_IF(timeout,
- "*** SERIOUS WARNING *** obtainBuffer() timed out "
- "but didn't need to be locked. We recovered, but "
- "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
-
audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
audioBuffer->channelCount = mChannelCount;
audioBuffer->frameCount = framesReq;
@@ -991,7 +1034,7 @@
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
- bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;
+ bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
}
// It is possible that we receive a flush()
// while the mixer is processing a block: in this case,
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index dacf75a..ba0d55b 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -15,6 +15,10 @@
** limitations under the License.
*/
+#define LOG_TAG "IAudioRecord"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
#include <sys/types.h>
@@ -42,8 +46,13 @@
{
Parcel data, reply;
data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
- remote()->transact(START, data, &reply);
- return reply.readInt32();
+ status_t status = remote()->transact(START, data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ } else {
+ LOGW("start() error: %s", strerror(-status));
+ }
+ return status;
}
virtual void stop()
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 7f43347..01ffd75 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -15,6 +15,10 @@
** limitations under the License.
*/
+#define LOG_TAG "IAudioTrack"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
#include <stdint.h>
#include <sys/types.h>
@@ -45,8 +49,13 @@
{
Parcel data, reply;
data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- remote()->transact(START, data, &reply);
- return reply.readInt32();
+ status_t status = remote()->transact(START, data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ } else {
+ LOGW("start() error: %s", strerror(-status));
+ }
+ return status;
}
virtual void stop()
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 91d0d00..60e3d71 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -798,7 +798,7 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-ToneGenerator::ToneGenerator(int streamType, float volume) {
+ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJava) {
LOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
@@ -808,6 +808,7 @@
LOGE("Unable to marshal AudioFlinger");
return;
}
+ mThreadCanCallJava = threadCanCallJava;
mStreamType = streamType;
mVolume = volume;
mpAudioTrack = 0;
@@ -1015,15 +1016,25 @@
}
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
- mpAudioTrack
- = new AudioTrack(mStreamType, 0, AudioSystem::PCM_16_BIT, AudioSystem::CHANNEL_OUT_MONO, 0, 0, audioCallback, this, 0);
-
+ mpAudioTrack = new AudioTrack();
if (mpAudioTrack == 0) {
LOGE("AudioTrack allocation failed");
goto initAudioTrack_exit;
}
LOGV("Create Track: %p\n", mpAudioTrack);
+ mpAudioTrack->set(mStreamType,
+ 0,
+ AudioSystem::PCM_16_BIT,
+ AudioSystem::CHANNEL_OUT_MONO,
+ 0,
+ 0,
+ audioCallback,
+ this,
+ 0,
+ 0,
+ mThreadCanCallJava);
+
if (mpAudioTrack->initCheck() != NO_ERROR) {
LOGE("AudioTrack->initCheck failed");
goto initAudioTrack_exit;