Merge "Camera2Api: Creating a JPEG stream needs to use the right size"
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b11c812..09160cc 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -128,8 +128,10 @@
// - BAD_VALUE: invalid parameter
// NOTE: this feature is not supported on all hardware platforms and it is
// necessary to check returned status before using the returned values.
- static status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
- audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getRenderPosition(audio_io_handle_t output,
+ size_t *halFrames,
+ size_t *dspFrames,
+ audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
// return the number of input frames lost by HAL implementation, or 0 if the handle is invalid
static size_t getInputFramesLost(audio_io_handle_t ioHandle);
@@ -197,7 +199,8 @@
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -245,6 +248,10 @@
static uint32_t getPrimaryOutputSamplingRate();
static size_t getPrimaryOutputFrameCount();
+ // Check if hw offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ static bool isOffloadSupported(const audio_offload_info_t& info);
+
// ----------------------------------------------------------------------------
private:
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e9bb76a..6727601 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -60,6 +60,8 @@
// Not currently used by android.media.AudioTrack.
EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and
// voluntary invalidation by mediaserver, or mediaserver crash.
+ EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played
+ // back (after stop is called)
};
/* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -175,7 +177,8 @@
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
@@ -198,7 +201,8 @@
void* user = NULL,
int notificationFrames = 0,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
@@ -233,7 +237,8 @@
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
int sessionId = 0,
- transfer_type transferType = TRANSFER_DEFAULT);
+ transfer_type transferType = TRANSFER_DEFAULT,
+ const audio_offload_info_t *offloadInfo = NULL);
/* Result of constructing the AudioTrack. This must be checked
* before using any AudioTrack API (except for set()), because using
@@ -521,6 +526,15 @@
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
public:
+//EL_FIXME to be reconciled with new obtainBuffer() return codes and control block proxy
+// enum {
+// NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
+// TEAR_DOWN = 0x80000002,
+// STOPPED = 1,
+// STREAM_END_WAIT,
+// STREAM_END
+// };
+
/* Release a filled buffer of "audioBuffer->frameCount" frames for AudioFlinger to process. */
// FIXME make private when obtainBuffer() for TRANSFER_OBTAIN is removed
void releaseBuffer(Buffer* audioBuffer);
@@ -550,6 +564,15 @@
*/
uint32_t getUnderrunFrames() const;
+ /* Get the flags */
+ audio_output_flags_t getFlags() const { return mFlags; }
+
+ /* Set parameters - only possible when using direct output */
+ status_t setParameters(const String8& keyValuePairs);
+
+ /* Get parameters */
+ String8 getParameters(const String8& keys);
+
protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
@@ -590,8 +613,11 @@
// NS_NEVER never again
static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3;
nsecs_t processAudioBuffer(const sp<AudioTrackThread>& thread);
+ status_t processStreamEnd(int32_t waitCount);
+
// caller must hold lock on mLock for all _l methods
+
status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
@@ -607,6 +633,8 @@
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
+ status_t getPosition_l(uint32_t *position);
+
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9c3067e..f8a9f2b 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -49,6 +49,7 @@
TRACK_DEFAULT = 0, // client requests a default AudioTrack
TRACK_TIMED = 1, // client requests a TimedAudioTrack
TRACK_FAST = 2, // client requests a fast AudioTrack or AudioRecord
+ TRACK_OFFLOAD = 4, // client requests offload to hw codec
};
typedef uint32_t track_flags_t;
@@ -137,7 +138,8 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags) = 0;
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2) = 0;
virtual status_t closeOutput(audio_io_handle_t output) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index b5ad4ef..09b9ea6 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -53,7 +53,8 @@
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = 0,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0) = 0;
@@ -95,6 +96,9 @@
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
uint32_t *count) = 0;
+ // Check if offload is possible for given format, stream type, sample rate,
+ // bit rate, duration, video and streaming or offload property is enabled
+ virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
};
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 144be0e..1014403 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -25,6 +25,7 @@
#include <binder/IInterface.h>
#include <binder/IMemory.h>
#include <utils/LinearTransform.h>
+#include <utils/String8.h>
namespace android {
@@ -82,6 +83,9 @@
or Tungsten time. The values for target are defined in AudioTrack.h */
virtual status_t setMediaTimeTransform(const LinearTransform& xform,
int target) = 0;
+
+ /* Send parameters to the audio hardware */
+ virtual status_t setParameters(const String8& keyValuePairs) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 9a75f81..61f7dc7 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -74,9 +74,18 @@
// AudioSink: abstraction layer for audio output
class AudioSink : public RefBase {
public:
+ enum cb_event_t {
+ CB_EVENT_FILL_BUFFER, // Request to write more data to buffer.
+ CB_EVENT_STREAM_END, // Sent after all the buffers queued in AF and HW are played
+ // back (after stop is called)
+ CB_EVENT_TEAR_DOWN // The AudioTrack was invalidated due to use case change:
+ // Need to re-evaluate offloading options
+ };
+
// Callback returns the number of bytes actually written to the buffer.
typedef size_t (*AudioCallback)(
- AudioSink *audioSink, void *buffer, size_t size, void *cookie);
+ AudioSink *audioSink, void *buffer, size_t size, void *cookie,
+ cb_event_t event);
virtual ~AudioSink() {}
virtual bool ready() const = 0; // audio output is open and ready
@@ -99,7 +108,8 @@
int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
AudioCallback cb = NULL,
void *cookie = NULL,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual void start() = 0;
virtual ssize_t write(const void* buffer, size_t size) = 0;
@@ -110,6 +120,9 @@
virtual status_t setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
virtual bool needsTrailingPadding() { return true; }
+
+ virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR; };
+ virtual String8 getParameters(const String8& keys) { return String8::empty(); };
};
MediaPlayerBase() : mCookie(0), mNotify(0) {}
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 3bf046d..ec9f2df 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -36,8 +36,13 @@
SEEK_COMPLETE
};
+ enum {
+ ALLOW_DEEP_BUFFERING = 0x01,
+ USE_OFFLOAD = 0x02
+ };
+
AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink,
- bool allowDeepBuffering = false,
+ uint32_t flags = 0,
AwesomePlayer *audioObserver = NULL);
virtual ~AudioPlayer();
@@ -67,6 +72,8 @@
status_t setPlaybackRatePermille(int32_t ratePermille);
+ void notifyAudioEOS();
+
private:
friend class VideoEditorAudioPlayer;
sp<MediaSource> mSource;
@@ -107,7 +114,8 @@
static size_t AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *data, size_t size, void *me);
+ void *data, size_t size, void *me,
+ MediaPlayerBase::AudioSink::cb_event_t event);
size_t fillBuffer(void *data, size_t size);
@@ -116,6 +124,7 @@
void reset();
uint32_t getNumFramesPendingPlayout() const;
+ int64_t getOutputPlayPositionUs_l() const;
AudioPlayer(const AudioPlayer &);
AudioPlayer &operator=(const AudioPlayer &);
diff --git a/include/media/stagefright/Utils.h b/include/media/stagefright/Utils.h
index 73940d3..c24f612 100644
--- a/include/media/stagefright/Utils.h
+++ b/include/media/stagefright/Utils.h
@@ -22,6 +22,8 @@
#include <stdint.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
+#include <system/audio.h>
+#include <media/MediaPlayerInterface.h>
namespace android {
@@ -48,6 +50,15 @@
AString MakeUserAgent();
+// Convert a MIME type to a AudioSystem::audio_format
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
+
+// Send information from MetaData to the HAL via AudioSink
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink, const sp<MetaData>& meta);
+
+// Check whether the stream defined by meta can be offloaded to hardware
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming);
+
} // namespace android
#endif // UTILS_H_
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ef5bb8d..b41684a 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -44,6 +44,10 @@
#define CBLK_BUFFER_END 0x80 // set by server when the position reaches end of buffer if not looping
#define CBLK_OVERRUN 0x100 // set by server immediately on input overrun, cleared by client
#define CBLK_INTERRUPT 0x200 // set by client on interrupt(), cleared by client in obtainBuffer()
+#define CBLK_STREAM_END_DONE 0x400 // set by server on render completion, cleared by client
+
+//EL_FIXME 20 seconds may not be enough and must be reconciled with new obtainBuffer implementation
+#define MAX_RUN_OFFLOADED_TIMEOUT_MS 20000 //assuming upto a maximum of 20 seconds of offloaded
struct AudioTrackSharedStreaming {
// similar to NBAIO MonoPipe
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
index 3fa8b87..dc360a5 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.cpp
@@ -575,10 +575,15 @@
size_t VideoEditorAudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *buffer, size_t size, void *cookie) {
+ void *buffer, size_t size, void *cookie,
+ MediaPlayerBase::AudioSink::cb_event_t event) {
VideoEditorAudioPlayer *me = (VideoEditorAudioPlayer *)cookie;
- return me->fillBuffer(buffer, size);
+ if (event == MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER ) {
+ return me->fillBuffer(buffer, size);
+ } else {
+ return 0;
+ }
}
diff --git a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
index a5616c1..d2e652d 100755
--- a/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorAudioPlayer.h
@@ -124,7 +124,8 @@
size_t fillBuffer(void *data, size_t size);
static size_t AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *data, size_t size, void *me);
+ void *data, size_t size, void *me,
+ MediaPlayerBase::AudioSink::cb_event_t event);
void reset();
void clear();
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.cpp b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
index 4a14b40..3384e34 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.cpp
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.cpp
@@ -391,7 +391,8 @@
status_t VideoEditorPlayer::VeAudioOutput::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t flags) {
+ AudioCallback cb, void *cookie, audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo) {
mCallback = cb;
mCallbackCookie = cookie;
@@ -545,7 +546,8 @@
AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
size_t actualSize = (*me->mCallback)(
- me, buffer->raw, buffer->size, me->mCallbackCookie);
+ me, buffer->raw, buffer->size, me->mCallbackCookie,
+ MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER);
buffer->size = actualSize;
diff --git a/libvideoeditor/lvpp/VideoEditorPlayer.h b/libvideoeditor/lvpp/VideoEditorPlayer.h
index defc90d..69323c3 100755
--- a/libvideoeditor/lvpp/VideoEditorPlayer.h
+++ b/libvideoeditor/lvpp/VideoEditorPlayer.h
@@ -52,7 +52,8 @@
virtual status_t open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t flags);
+ AudioCallback cb, void *cookie, audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
virtual void start();
virtual ssize_t write(const void* buffer, size_t size);
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 693df60..a6dedec 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -361,8 +361,8 @@
return af->setVoiceVolume(value);
}
-status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames,
- audio_stream_type_t stream)
+status_t AudioSystem::getRenderPosition(audio_io_handle_t output, size_t *halFrames,
+ size_t *dspFrames, audio_stream_type_t stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -371,7 +371,11 @@
stream = AUDIO_STREAM_MUSIC;
}
- return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
+ if (output == 0) {
+ output = getOutput(stream);
+ }
+
+ return af->getRenderPosition(halFrames, dspFrames, output);
}
size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -585,11 +589,12 @@
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getOutput(stream, samplingRate, format, channelMask, flags);
+ return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
@@ -771,6 +776,14 @@
gOutputs.clear();
}
+bool AudioSystem::isOffloadSupported(const audio_offload_info_t& info)
+{
+ ALOGV("isOffloadSupported()");
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return false;
+ return aps->isOffloadSupported(info);
+}
+
// ---------------------------------------------------------------------------
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index faca054..2af162c 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -97,7 +97,8 @@
void* user,
int notificationFrames,
int sessionId,
- transfer_type transferType)
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -105,7 +106,7 @@
{
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
- 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType);
+ 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
}
AudioTrack::AudioTrack(
@@ -119,7 +120,8 @@
void* user,
int notificationFrames,
int sessionId,
- transfer_type transferType)
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -127,7 +129,7 @@
{
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
- sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType);
+ sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo);
}
AudioTrack::~AudioTrack()
@@ -164,7 +166,8 @@
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId,
- transfer_type transferType)
+ transfer_type transferType,
+ const audio_offload_info_t *offloadInfo)
{
switch (transferType) {
case TRANSFER_DEFAULT:
@@ -284,7 +287,8 @@
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
- flags);
+ flags,
+ offloadInfo);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
@@ -1543,6 +1547,21 @@
return result;
}
+status_t AudioTrack::setParameters(const String8& keyValuePairs)
+{
+ AutoMutex lock(mLock);
+ if (mAudioTrack != 0) {
+ return mAudioTrack->setParameters(keyValuePairs);
+ } else {
+ return NO_INIT;
+ }
+}
+
+String8 AudioTrack::getParameters(const String8& keys)
+{
+ return String8::empty();
+}
+
status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
{
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 2f18680..e4df77d 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -361,7 +361,8 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
Parcel data, reply;
audio_devices_t devices = pDevices ? *pDevices : (audio_devices_t)0;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 386c351..57de58f 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -56,7 +56,8 @@
GET_DEVICES_FOR_STREAM,
QUERY_DEFAULT_PRE_PROCESSING,
SET_EFFECT_ENABLED,
- IS_STREAM_ACTIVE_REMOTELY
+ IS_STREAM_ACTIVE_REMOTELY,
+ IS_OFFLOAD_SUPPORTED
};
class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -126,7 +127,8 @@
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -374,6 +376,12 @@
*count = retCount;
return status;
}
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& info)
+ {
+ // stub function
+ return false;
+ }
};
IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index e92f8aa..a2b49a3 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -39,6 +39,7 @@
ALLOCATE_TIMED_BUFFER,
QUEUE_TIMED_BUFFER,
SET_MEDIA_TIME_TRANSFORM,
+ SET_PARAMETERS
};
class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -154,6 +155,17 @@
}
return status;
}
+
+ virtual status_t setParameters(const String8& keyValuePairs) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeString8(keyValuePairs);
+ status_t status = remote()->transact(SET_PARAMETERS, data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -223,6 +235,12 @@
reply->writeInt32(setMediaTimeTransform(xform, target));
return NO_ERROR;
} break;
+ case SET_PARAMETERS: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ String8 keyValuePairs(data.readString8());
+ reply->writeInt32(setParameters(keyValuePairs));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index fa1ff36..afde373 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -321,7 +321,7 @@
mHeap->getBase(), mHeap->getSize(), mHeap->getFlags(), mHeap->getDevice());
result.append(buffer);
}
- snprintf(buffer, 255, " msec per frame(%f), channel count(%d), format(%d), frame count(%ld)\n",
+ snprintf(buffer, 255, " msec per frame(%f), channel count(%d), format(%d), frame count(%zd)\n",
mMsecsPerFrame, mChannelCount, mFormat, mFrameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), size(%d), error(%d), command complete(%s)\n",
@@ -1385,7 +1385,8 @@
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
mCallback = cb;
mCallbackCookie = cookie;
@@ -1661,7 +1662,8 @@
}
size_t actualSize = (*me->mCallback)(
- me, buffer->raw, buffer->size, me->mCallbackCookie);
+ me, buffer->raw, buffer->size, me->mCallbackCookie,
+ CB_EVENT_FILL_BUFFER);
if (actualSize == 0 && buffer->size > 0 && me->mNextOutput == NULL) {
// We've reached EOS but the audio track is not stopped yet,
@@ -1767,7 +1769,8 @@
}
size_t actualSize =
- (*mCallback)(sink.get(), mBuffer, mBufferSize, mCookie);
+ (*mCallback)(sink.get(), mBuffer, mBufferSize, mCookie,
+ MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER);
if (actualSize > 0) {
sink->write(mBuffer, actualSize);
@@ -1781,7 +1784,8 @@
status_t MediaPlayerService::AudioCache::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t flags)
+ AudioCallback cb, void *cookie, audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
if (mHeap->getHeapID() < 0) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index e586156..1f8bcc7 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -94,7 +94,8 @@
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
virtual void start();
virtual ssize_t write(const void* buffer, size_t size);
@@ -195,7 +196,8 @@
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount = 1,
AudioCallback cb = NULL, void *cookie = NULL,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
virtual void start();
virtual ssize_t write(const void* buffer, size_t size);
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 92efae8..61d6746 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -34,7 +34,7 @@
AudioPlayer::AudioPlayer(
const sp<MediaPlayerBase::AudioSink> &audioSink,
- bool allowDeepBuffering,
+ uint32_t flags,
AwesomePlayer *observer)
: mInputBuffer(NULL),
mSampleRate(0),
@@ -52,7 +52,7 @@
mFirstBufferResult(OK),
mFirstBuffer(NULL),
mAudioSink(audioSink),
- mAllowDeepBuffering(allowDeepBuffering),
+ mAllowDeepBuffering((flags & ALLOW_DEEP_BUFFERING) != 0),
mObserver(observer),
mPinnedTimeUs(-1ll) {
}
@@ -304,7 +304,8 @@
// static
size_t AudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
- void *buffer, size_t size, void *cookie) {
+ void *buffer, size_t size, void *cookie,
+ MediaPlayerBase::AudioSink::cb_event_t event) {
AudioPlayer *me = (AudioPlayer *)cookie;
return me->fillBuffer(buffer, size);
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index a01ec97..ae6ae2d 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -282,6 +282,7 @@
if (!usesReservoir()) {
if (u.ext_data) {
free(u.ext_data);
+ u.ext_data = NULL;
}
}
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index b0df379..e9789d3 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -471,5 +471,24 @@
return ua;
}
+status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink,
+ const sp<MetaData>& meta)
+{
+ // stub
+ return OK;
+}
+
+status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime)
+{
+ // stub
+ return BAD_VALUE;
+}
+
+bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo, bool isStreaming)
+{
+ // stub
+ return false;
+}
+
} // namespace android
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 2306f31..0d17d65 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -25,6 +25,7 @@
#include <media/stagefright/DataSource.h>
#include <media/stagefright/OMXClient.h>
#include <media/stagefright/TimeSource.h>
+#include <media/stagefright/MetaData.h>
#include <utils/threads.h>
#include <drm/DrmManagerClient.h>
@@ -100,7 +101,7 @@
void postAudioEOS(int64_t delayUs = 0ll);
void postAudioSeekComplete();
-
+ void postAudioTearDown();
status_t dump(int fd, const Vector<String16> &args) const;
private:
@@ -171,6 +172,7 @@
ssize_t mActiveAudioTrackIndex;
sp<MediaSource> mAudioTrack;
+ sp<MediaSource> mOmxSource;
sp<MediaSource> mAudioSource;
AudioPlayer *mAudioPlayer;
int64_t mDurationUs;
@@ -211,7 +213,8 @@
bool mAudioStatusEventPending;
sp<TimedEventQueue::Event> mVideoLagEvent;
bool mVideoLagEventPending;
-
+ sp<TimedEventQueue::Event> mAudioTearDownEvent;
+ bool mAudioTearDownEventPending;
sp<TimedEventQueue::Event> mAsyncPrepareEvent;
Condition mPreparedCondition;
bool mIsAsyncPrepare;
@@ -223,6 +226,8 @@
void postStreamDoneEvent_l(status_t status);
void postCheckAudioStatusEvent(int64_t delayUs);
void postVideoLagEvent_l();
+ void postAudioTearDownEvent();
+
status_t play_l();
MediaBuffer *mVideoBuffer;
@@ -257,6 +262,7 @@
void setAudioSource(sp<MediaSource> source);
status_t initAudioDecoder();
+
void setVideoSource(sp<MediaSource> source);
status_t initVideoDecoder(uint32_t flags = 0);
@@ -273,6 +279,9 @@
void abortPrepare(status_t err);
void finishAsyncPrepare_l();
void onVideoLagUpdate();
+ void onAudioTearDownEvent();
+
+ void beginPrepareAsync_l();
bool getCachedDuration_l(int64_t *durationUs, bool *eos);
@@ -285,6 +294,7 @@
void finishSeekIfNecessary(int64_t videoTimeUs);
void ensureCacheIsFetching_l();
+ void createAudioPlayer_l();
status_t startAudioPlayer_l(bool sendErrorNotification = true);
void shutdownVideoDecoder_l();
@@ -327,6 +337,9 @@
Vector<TrackStat> mTracks;
} mStats;
+ bool mOffloadAudio;
+ bool mAudioTearDown;
+
status_t setVideoScalingMode(int32_t mode);
status_t setVideoScalingMode_l(int32_t mode);
status_t getTrackInfo(Parcel* reply) const;
diff --git a/media/libstagefright/include/ESDS.h b/media/libstagefright/include/ESDS.h
index 3a79951..2f40dae 100644
--- a/media/libstagefright/include/ESDS.h
+++ b/media/libstagefright/include/ESDS.h
@@ -33,6 +33,9 @@
status_t getObjectTypeIndication(uint8_t *objectTypeIndication) const;
status_t getCodecSpecificInfo(const void **data, size_t *size) const;
+ status_t getCodecSpecificOffset(size_t *offset, size_t *size) const;
+ status_t getBitRate(uint32_t *brateMax, uint32_t *brateAvg) const;
+ status_t getStreamType(uint8_t *streamType) const;
private:
enum {
@@ -49,6 +52,9 @@
size_t mDecoderSpecificOffset;
size_t mDecoderSpecificLength;
uint8_t mObjectTypeIndication;
+ uint8_t mStreamType;
+ uint32_t mBitRateMax;
+ uint32_t mBitRateAvg;
status_t skipDescriptorHeader(
size_t offset, size_t size,
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index a6edb77..c8e8aba 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -981,11 +981,12 @@
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
- struct audio_config config = {
- sample_rate: sampleRate,
- channel_mask: channelMask,
- format: format,
- };
+ struct audio_config config;
+ memset(&config, 0, sizeof(config));
+ config.sample_rate = sampleRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
size_t size = dev->get_input_buffer_size(dev, &config);
mHardwareStatus = AUDIO_HW_IDLE;
@@ -1388,15 +1389,19 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
status_t status;
PlaybackThread *thread = NULL;
- struct audio_config config = {
- sample_rate: pSamplingRate ? *pSamplingRate : 0,
- channel_mask: pChannelMask ? *pChannelMask : 0,
- format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
- };
+ struct audio_config config;
+ config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
+ config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
+ config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
+ if (offloadInfo) {
+ config.offload_info = *offloadInfo;
+ }
+
audio_stream_out_t *outStream = NULL;
AudioHwDevice *outHwDev;
@@ -1591,11 +1596,11 @@
{
status_t status;
RecordThread *thread = NULL;
- struct audio_config config = {
- sample_rate: pSamplingRate ? *pSamplingRate : 0,
- channel_mask: pChannelMask ? *pChannelMask : 0,
- format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
- };
+ struct audio_config config;
+ config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
+ config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
+ config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
+
uint32_t reqSamplingRate = config.sample_rate;
audio_format_t reqFormat = config.format;
audio_channel_mask_t reqChannels = config.channel_mask;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 05dbab1..b640b31 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -160,7 +160,8 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags);
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2);
@@ -406,6 +407,8 @@
int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
+
+ virtual status_t setParameters(const String8& keyValuePairs);
private:
const sp<PlaybackThread::Track> mTrack;
};
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 2706880..fd4431c 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -222,15 +222,16 @@
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
if (mpAudioPolicy == NULL) {
return 0;
}
ALOGV("getOutput()");
Mutex::Autolock _l(mLock);
- return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask,
- flags);
+ return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate,
+ format, channelMask, flags, offloadInfo);
}
status_t AudioPolicyService::startOutput(audio_io_handle_t output,
@@ -1055,6 +1056,11 @@
return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
}
+bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
+{
+ return false; // stub function
+}
+
// ----------------------------------------------------------------------------
// Audio pre-processing configuration
// ----------------------------------------------------------------------------
@@ -1387,7 +1393,8 @@
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
- audio_output_flags_t flags)
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 53238fa..e723c47 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -67,7 +67,8 @@
audio_format_t format = AUDIO_FORMAT_DEFAULT,
audio_channel_mask_t channelMask = 0,
audio_output_flags_t flags =
- AUDIO_OUTPUT_FLAG_NONE);
+ AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -136,6 +137,7 @@
virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream);
virtual status_t stopTone();
virtual status_t setVoiceVolume(float volume, int delayMs = 0);
+ virtual bool isOffloadSupported(const audio_offload_info_t &config);
private:
AudioPolicyService() ANDROID_API;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index bfc197c..f0dbee3 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -250,6 +250,10 @@
mTrack->pause();
}
+status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
+ return INVALID_OPERATION; // stub function
+}
+
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
diff --git a/services/camera/libcameraservice/Camera2Client.cpp b/services/camera/libcameraservice/Camera2Client.cpp
index a94c658..96bde90 100644
--- a/services/camera/libcameraservice/Camera2Client.cpp
+++ b/services/camera/libcameraservice/Camera2Client.cpp
@@ -595,6 +595,21 @@
void Camera2Client::setPreviewCallbackFlagL(Parameters ¶ms, int flag) {
status_t res = OK;
+
+ switch(params.state) {
+ case Parameters::STOPPED:
+ case Parameters::WAITING_FOR_PREVIEW_WINDOW:
+ case Parameters::PREVIEW:
+ // OK
+ break;
+ default:
+ if (flag & CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK) {
+ ALOGE("%s: Camera %d: Can't use preview callbacks "
+ "in state %d", __FUNCTION__, mCameraId, params.state);
+ return;
+ }
+ }
+
if (flag & CAMERA_FRAME_CALLBACK_FLAG_ONE_SHOT_MASK) {
ALOGV("%s: setting oneshot", __FUNCTION__);
params.previewCallbackOneShot = true;
@@ -615,24 +630,15 @@
params.previewCallbackFlags = flag;
- switch(params.state) {
- case Parameters::PREVIEW:
- res = startPreviewL(params, true);
- break;
- case Parameters::RECORD:
- case Parameters::VIDEO_SNAPSHOT:
- res = startRecordingL(params, true);
- break;
- default:
- break;
- }
- if (res != OK) {
- ALOGE("%s: Camera %d: Unable to refresh request in state %s",
- __FUNCTION__, mCameraId,
- Parameters::getStateName(params.state));
+ if (params.state == Parameters::PREVIEW) {
+ res = startPreviewL(params, true);
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to refresh request in state %s",
+ __FUNCTION__, mCameraId,
+ Parameters::getStateName(params.state));
+ }
}
}
-
}
status_t Camera2Client::setPreviewCallbackTarget(
@@ -755,6 +761,26 @@
params.previewCallbackSurface;
if (callbacksEnabled) {
+ // Can't have recording stream hanging around when enabling callbacks,
+ // since it exceeds the max stream count on some devices.
+ if (mStreamingProcessor->getRecordingStreamId() != NO_STREAM) {
+ ALOGV("%s: Camera %d: Clearing out recording stream before "
+ "creating callback stream", __FUNCTION__, mCameraId);
+ res = mStreamingProcessor->stopStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Can't stop streaming to delete "
+ "recording stream", __FUNCTION__, mCameraId);
+ return res;
+ }
+ res = mStreamingProcessor->deleteRecordingStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete recording stream before "
+ "enabling callbacks: %s (%d)", __FUNCTION__, mCameraId,
+ strerror(-res), res);
+ return res;
+ }
+ }
+
res = mCallbackProcessor->updateStream(params);
if (res != OK) {
ALOGE("%s: Camera %d: Unable to update callback stream: %s (%d)",
@@ -951,6 +977,29 @@
}
}
+ // Not all devices can support a preview callback stream and a recording
+ // stream at the same time, so assume none of them can.
+ if (mCallbackProcessor->getStreamId() != NO_STREAM) {
+ ALOGV("%s: Camera %d: Clearing out callback stream before "
+ "creating recording stream", __FUNCTION__, mCameraId);
+ res = mStreamingProcessor->stopStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Can't stop streaming to delete callback stream",
+ __FUNCTION__, mCameraId);
+ return res;
+ }
+ res = mCallbackProcessor->deleteStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete callback stream before "
+ "record: %s (%d)", __FUNCTION__, mCameraId,
+ strerror(-res), res);
+ return res;
+ }
+ }
+ // Disable callbacks if they're enabled; can't record and use callbacks,
+ // and we can't fail record start without stagefright asserting.
+ params.previewCallbackFlags = 0;
+
res = updateProcessorStream<
StreamingProcessor,
&StreamingProcessor::updateRecordingStream>(mStreamingProcessor,
@@ -962,19 +1011,6 @@
}
Vector<uint8_t> outputStreams;
- bool callbacksEnabled = (params.previewCallbackFlags &
- CAMERA_FRAME_CALLBACK_FLAG_ENABLE_MASK) ||
- params.previewCallbackSurface;
-
- if (callbacksEnabled) {
- res = mCallbackProcessor->updateStream(params);
- if (res != OK) {
- ALOGE("%s: Camera %d: Unable to update callback stream: %s (%d)",
- __FUNCTION__, mCameraId, strerror(-res), res);
- return res;
- }
- outputStreams.push(getCallbackStreamId());
- }
outputStreams.push(getPreviewStreamId());
outputStreams.push(getRecordingStreamId());
@@ -1706,6 +1742,8 @@
* queue) and then try again. Resume streaming once we're done.
*/
if (res == -EBUSY) {
+ ALOGV("%s: Camera %d: Pausing to update stream", __FUNCTION__,
+ mCameraId);
res = mStreamingProcessor->togglePauseStream(/*pause*/true);
if (res != OK) {
ALOGE("%s: Camera %d: Can't pause streaming: %s (%d)",
diff --git a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
index bc81409..aae2504 100644
--- a/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
+++ b/services/camera/libcameraservice/camera2/CallbackProcessor.cpp
@@ -109,7 +109,9 @@
if (!mCallbackToApp && mCallbackConsumer == 0) {
// Create CPU buffer queue endpoint, since app hasn't given us one
- mCallbackConsumer = new CpuConsumer(kCallbackHeapCount);
+ // Make it async to avoid disconnect deadlocks
+ mCallbackConsumer = new CpuConsumer(kCallbackHeapCount,
+ /*synchronized*/ false);
mCallbackConsumer->setFrameAvailableListener(this);
mCallbackConsumer->setName(String8("Camera2Client::CallbackConsumer"));
mCallbackWindow = new Surface(
@@ -167,7 +169,7 @@
status_t CallbackProcessor::deleteStream() {
ATRACE_CALL();
sp<CameraDeviceBase> device;
-
+ status_t res;
{
Mutex::Autolock l(mInputMutex);
@@ -180,7 +182,19 @@
return INVALID_OPERATION;
}
}
- device->deleteStream(mCallbackStreamId);
+ res = device->waitUntilDrained();
+ if (res != OK) {
+ ALOGE("%s: Error waiting for HAL to drain: %s (%d)",
+ __FUNCTION__, strerror(-res), res);
+ return res;
+ }
+
+ res = device->deleteStream(mCallbackStreamId);
+ if (res != OK) {
+ ALOGE("%s: Unable to delete callback stream: %s (%d)",
+ __FUNCTION__, strerror(-res), res);
+ return res;
+ }
{
Mutex::Autolock l(mInputMutex);