Upintegrate Audio Flinger changes from ICS_AAH
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index aa0b8f8..9396a0b 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -22,11 +22,14 @@
#include <sys/types.h>
#include <limits.h>
+#include <common_time/cc_helper.h>
+
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
#include <utils/Atomic.h>
#include <utils/Errors.h>
@@ -55,7 +58,7 @@
// ----------------------------------------------------------------------------
-static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
public BinderService<AudioFlinger>,
@@ -78,6 +81,7 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status);
@@ -102,6 +106,7 @@
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
+ virtual float masterVolumeSW() const;
virtual bool masterMute() const;
virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
@@ -206,6 +211,8 @@
audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
void purgeStaleEffects_l();
+ static nsecs_t mStandbyTimeInNsecs;
+
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
status_t dumpClients(int fd, const Vector<String16>& args);
@@ -220,12 +227,18 @@
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
+ bool reserveTimedTrack();
+ void releaseTimedTrack();
+
private:
Client(const Client&);
Client& operator = (const Client&);
const sp<AudioFlinger> mAudioFlinger;
const sp<MemoryDealer> mMemoryDealer;
const pid_t mPid;
+
+ Mutex mTimedTrackLock;
+ int mTimedTrackCount;
};
// --- Notification Client ---
@@ -333,7 +346,9 @@
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
audio_format_t format() const {
@@ -609,7 +624,6 @@
int16_t *mainBuffer() const { return mMainBuffer; }
int auxEffectId() const { return mAuxEffectId; }
-
protected:
friend class ThreadBase;
friend class TrackHandle;
@@ -620,7 +634,11 @@
Track(const Track&);
Track& operator = (const Track&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual uint32_t framesReady() const;
+
bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
@@ -636,6 +654,8 @@
return (mStreamType == AUDIO_STREAM_CNT);
}
+ virtual bool isTimedTrack() const { return false; }
+
// we don't really need a lock for these
volatile bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
@@ -652,6 +672,79 @@
bool mHasVolumeController;
}; // end of Track
+ class TimedTrack : public Track {
+ public:
+ static sp<TimedTrack> create(const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+ ~TimedTrack();
+
+ class TimedBuffer {
+ public:
+ TimedBuffer();
+ TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+ const sp<IMemory>& buffer() const { return mBuffer; }
+ int64_t pts() const { return mPTS; }
+ int position() const { return mPosition; }
+ void setPosition(int pos) { mPosition = pos; }
+ private:
+ sp<IMemory> mBuffer;
+ int64_t mPTS;
+ int mPosition;
+ };
+
+ virtual bool isTimedTrack() const { return true; }
+
+ virtual uint32_t framesReady() const;
+
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSilence(uint32_t numFrames,
+ AudioBufferProvider::Buffer* buffer);
+
+ status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TimedAudioTrack::TargetTimeline target);
+ void trimTimedBufferQueue_l();
+
+ private:
+ TimedTrack(const wp<ThreadBase>& thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+
+ uint64_t mLocalTimeFreq;
+ LinearTransform mLocalTimeToSampleTransform;
+ sp<MemoryDealer> mTimedMemoryDealer;
+ Vector<TimedBuffer> mTimedBufferQueue;
+ uint8_t* mTimedSilenceBuffer;
+ uint32_t mTimedSilenceBufferSize;
+ mutable Mutex mTimedBufferQueueLock;
+ bool mTimedAudioOutputOnTime;
+ CCHelper mCCHelper;
+
+ Mutex mMediaTimeTransformLock;
+ LinearTransform mMediaTimeTransform;
+ bool mMediaTimeTransformValid;
+ TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+ };
+
// playback track
class OutputTrack : public Track {
@@ -726,6 +819,7 @@
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status);
AudioStreamOut* getOutput() const;
@@ -920,6 +1014,12 @@
virtual void mute(bool);
virtual void pause();
virtual status_t attachAuxEffect(int effectId);
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
@@ -967,7 +1067,9 @@
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
bool mOverflow;
};
@@ -1004,7 +1106,8 @@
AudioStreamIn* clearInput();
virtual audio_stream_t* stream();
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
@@ -1401,6 +1504,28 @@
friend class RecordThread;
friend class PlaybackThread;
+ enum master_volume_support {
+ // MVS_NONE:
+ // Audio HAL has no support for master volume, either setting or
+ // getting. All master volume control must be implemented in SW by the
+ // AudioFlinger mixing core.
+ MVS_NONE,
+
+ // MVS_SETONLY:
+ // Audio HAL has support for setting master volume, but not for getting
+ // master volume (original HAL design did not include a getter).
+ // AudioFlinger needs to keep track of the last set master volume in
+ // addition to needing to set an initial, default, master volume at HAL
+ // load time.
+ MVS_SETONLY,
+
+ // MVS_FULL:
+ // Audio HAL has support both for setting and getting master volume.
+ // AudioFlinger should send all set and get master volume requests
+ // directly to the HAL.
+ MVS_FULL,
+ };
+
mutable Mutex mLock;
DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
@@ -1429,6 +1554,7 @@
AUDIO_SET_VOICE_VOLUME,
AUDIO_SET_PARAMETER,
AUDIO_HW_GET_INPUT_BUFFER_SIZE,
+ AUDIO_HW_GET_MASTER_VOLUME,
};
mutable hardware_call_state mHardwareStatus; // for dump only
@@ -1439,6 +1565,8 @@
// both are protected by mLock
float mMasterVolume;
+ float mMasterVolumeSW;
+ master_volume_support mMasterVolumeSupportLvl;
bool mMasterMute;
DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
@@ -1451,7 +1579,8 @@
// protected by mLock
Vector<AudioSessionRef*> mAudioSessionRefs;
- float masterVolume_l() const { return mMasterVolume; }
+ float masterVolume_l() const;
+ float masterVolumeSW_l() const { return mMasterVolumeSW; }
bool masterMute_l() const { return mMasterMute; }
private: