summaryrefslogtreecommitdiffstats
path: root/libs/audioflinger
diff options
context:
space:
mode:
Diffstat (limited to 'libs/audioflinger')
-rw-r--r--libs/audioflinger/A2dpAudioInterface.cpp4
-rw-r--r--libs/audioflinger/A2dpAudioInterface.h1
-rw-r--r--libs/audioflinger/AudioBufferProvider.h2
-rw-r--r--libs/audioflinger/AudioDumpInterface.h6
-rw-r--r--libs/audioflinger/AudioFlinger.cpp115
-rw-r--r--libs/audioflinger/AudioFlinger.h20
-rw-r--r--libs/audioflinger/AudioHardwareGeneric.cpp11
-rw-r--r--libs/audioflinger/AudioHardwareGeneric.h1
-rw-r--r--libs/audioflinger/AudioHardwareInterface.cpp2
-rw-r--r--libs/audioflinger/AudioHardwareStub.cpp9
-rw-r--r--libs/audioflinger/AudioHardwareStub.h1
11 files changed, 87 insertions, 85 deletions
diff --git a/libs/audioflinger/A2dpAudioInterface.cpp b/libs/audioflinger/A2dpAudioInterface.cpp
index b6d5078..16a4f2d 100644
--- a/libs/audioflinger/A2dpAudioInterface.cpp
+++ b/libs/audioflinger/A2dpAudioInterface.cpp
@@ -71,8 +71,8 @@ AudioStreamOut* A2dpAudioInterface::openOutputStream(
}
AudioStreamIn* A2dpAudioInterface::openInputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status,
- AudioSystem::audio_in_acoustics acoustics)
+ int inputSource, int format, int channelCount, uint32_t sampleRate,
+ status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
if (status)
*status = -1;
diff --git a/libs/audioflinger/A2dpAudioInterface.h b/libs/audioflinger/A2dpAudioInterface.h
index 7901a8c..091e775 100644
--- a/libs/audioflinger/A2dpAudioInterface.h
+++ b/libs/audioflinger/A2dpAudioInterface.h
@@ -55,6 +55,7 @@ public:
status_t *status=0);
virtual AudioStreamIn* openInputStream(
+ int inputSource,
int format,
int channelCount,
uint32_t sampleRate,
diff --git a/libs/audioflinger/AudioBufferProvider.h b/libs/audioflinger/AudioBufferProvider.h
index 1a467c7..81c5c39 100644
--- a/libs/audioflinger/AudioBufferProvider.h
+++ b/libs/audioflinger/AudioBufferProvider.h
@@ -36,6 +36,8 @@ public:
};
size_t frameCount;
};
+
+ virtual ~AudioBufferProvider() {}
virtual status_t getNextBuffer(Buffer* buffer) = 0;
virtual void releaseBuffer(Buffer* buffer) = 0;
diff --git a/libs/audioflinger/AudioDumpInterface.h b/libs/audioflinger/AudioDumpInterface.h
index 9a94102..b72c94e 100644
--- a/libs/audioflinger/AudioDumpInterface.h
+++ b/libs/audioflinger/AudioDumpInterface.h
@@ -78,9 +78,9 @@ public:
virtual status_t setParameter(const char* key, const char* value)
{return mFinalInterface->setParameter(key, value);}
- virtual AudioStreamIn* openInputStream( int format, int channelCount, uint32_t sampleRate, status_t *status,
- AudioSystem::audio_in_acoustics acoustics)
- {return mFinalInterface->openInputStream( format, channelCount, sampleRate, status, acoustics);}
+ virtual AudioStreamIn* openInputStream(int inputSource, int format, int channelCount,
+ uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
+ { return mFinalInterface->openInputStream(inputSource, format, channelCount, sampleRate, status, acoustics); }
virtual status_t dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 43df7dd..2817a0d 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -499,7 +499,8 @@ status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
}
#ifdef WITH_A2DP
- LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid());
+ LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
+ IPCThreadState::self()->getCallingPid());
if (mode == AudioSystem::MODE_NORMAL &&
(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
AutoMutex lock(&mLock);
@@ -655,16 +656,12 @@ status_t AudioFlinger::setStreamVolume(int stream, float value)
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
- float hwValue = value;
+ float hwValue;
if (stream == AudioSystem::VOICE_CALL) {
hwValue = (float)AudioSystem::logToLinear(value)/100.0f;
- // FIXME: This is a temporary fix to re-base the internally
- // generated in-call audio so that it is never muted, which is
- // already the case for the hardware routed in-call audio.
- // When audio stream handling is reworked, this should be
- // addressed more cleanly. Fixes #1324; see discussion at
- // http://review.source.android.com/8224
- value = value * 0.99 + 0.01;
+ // offset value to reflect actual hardware volume that never reaches 0
+ // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
+ value = 0.01 + 0.99 * value;
} else { // (type == AudioSystem::BLUETOOTH_SCO)
hwValue = 1.0f;
}
@@ -681,6 +678,11 @@ status_t AudioFlinger::setStreamVolume(int stream, float value)
mA2dpMixerThread->setStreamVolume(stream, value);
#endif
+ mHardwareMixerThread->setStreamVolume(stream, value);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setStreamVolume(stream, value);
+#endif
+
return ret;
}
@@ -718,15 +720,14 @@ float AudioFlinger::streamVolume(int stream) const
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return 0.0f;
}
- float value = mHardwareMixerThread->streamVolume(stream);
+ float volume = mHardwareMixerThread->streamVolume(stream);
+ // remove correction applied by setStreamVolume()
if (stream == AudioSystem::VOICE_CALL) {
- // FIXME: Re-base internally generated in-call audio,
- // reverse of above in setStreamVolume.
- value = (value - 0.01) / 0.99;
+ volume = (volume - 0.01) / 0.99 ;
}
- return value;
+ return volume;
}
bool AudioFlinger::streamMute(int stream) const
@@ -824,24 +825,22 @@ void AudioFlinger::handleForcedSpeakerRoute(int command)
{
AutoMutex lock(mHardwareLock);
if (mForcedSpeakerCount++ == 0) {
- mRouteRestoreTime = 0;
- mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
- if (mForcedRoute == 0 && !(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
- LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- mAudioHardware->setMasterVolume(0);
- usleep(mHardwareMixerThread->latency()*1000);
- mHardwareStatus = AUDIO_HW_SET_ROUTING;
- mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareStatus = AUDIO_HW_IDLE;
- // delay track start so that audio hardware has time to siwtch routes
- usleep(kStartSleepTime);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- mAudioHardware->setMasterVolume(mHardwareMixerThread->masterVolume());
- mHardwareStatus = AUDIO_HW_IDLE;
+ if (mForcedRoute == 0) {
+ mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
+ LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
+ if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+ LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
+ usleep(mHardwareMixerThread->latency()*1000);
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ // delay track start so that audio hardware has time to siwtch routes
+ usleep(kStartSleepTime);
+ }
}
mForcedRoute = AudioSystem::ROUTE_SPEAKER;
+ mRouteRestoreTime = 0;
}
LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
}
@@ -902,7 +901,7 @@ void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
}
LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
} else {
- LOGE("mA2dpDisableCount is already zero");
+ LOGV("mA2dpDisableCount is already zero");
}
}
}
@@ -1289,7 +1288,7 @@ sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
status_t lStatus;
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
- if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
+ if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
@@ -1497,18 +1496,6 @@ status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
return status;
}
-// removeTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::removeTrack_l(wp<Track> track, int name)
-{
- sp<Track> t = track.promote();
- if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
- t->reset();
- deleteTrackName_l(name);
- removeActiveTrack_l(track);
- mAudioFlinger->mWaitWorkCV.broadcast();
- }
-}
-
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
{
@@ -1577,7 +1564,6 @@ size_t AudioFlinger::MixerThread::getOutputFrameCount()
AudioFlinger::MixerThread::TrackBase::TrackBase(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
uint32_t sampleRate,
int format,
int channelCount,
@@ -1587,7 +1573,6 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
: RefBase(),
mMixerThread(mixerThread),
mClient(client),
- mStreamType(streamType),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
@@ -1618,8 +1603,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
- mCblk->sampleRate = (uint16_t)sampleRate;
- mCblk->channels = (uint16_t)channelCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -1642,8 +1627,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
- mCblk->sampleRate = (uint16_t)sampleRate;
- mCblk->channels = (uint16_t)channelCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
@@ -1704,7 +1689,7 @@ int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
}
int AudioFlinger::MixerThread::TrackBase::channelCount() const {
- return mCblk->channels;
+ return (int)mCblk->channels;
}
void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -1714,7 +1699,7 @@ void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- cblk->channels == 2 && ((unsigned long)bufferStart & 3) ) {
+ (cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channels %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
@@ -1737,12 +1722,13 @@ AudioFlinger::MixerThread::Track::Track(
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
- : TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+ : TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
{
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
mMute = false;
mSharedBuffer = sharedBuffer;
+ mStreamType = streamType;
}
AudioFlinger::MixerThread::Track::~Track()
@@ -1750,7 +1736,6 @@ AudioFlinger::MixerThread::Track::~Track()
wp<Track> weak(this); // never create a strong ref from the dtor
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mState = TERMINATED;
- mMixerThread->removeTrack_l(weak, mName);
}
void AudioFlinger::MixerThread::Track::destroy()
@@ -1927,15 +1912,15 @@ void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
AudioFlinger::MixerThread::RecordTrack::RecordTrack(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags)
- : TrackBase(mixerThread, client, streamType, sampleRate, format,
+ : TrackBase(mixerThread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
- mOverflow(false)
+ mOverflow(false), mInputSource(inputSource)
{
}
@@ -2260,7 +2245,7 @@ status_t AudioFlinger::TrackHandle::onTransact(
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
@@ -2283,18 +2268,12 @@ sp<IAudioRecord> AudioFlinger::openRecord(
goto Exit;
}
- if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+ if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
- if (sampleRate > MAX_SAMPLE_RATE) {
- LOGE("Sample rate out of range");
- lStatus = BAD_VALUE;
- goto Exit;
- }
-
if (mAudioRecordThread == 0) {
LOGE("Audio record thread not started");
lStatus = NO_INIT;
@@ -2326,7 +2305,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
- recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, streamType, sampleRate,
+ recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
format, channelCount, frameCount, flags);
}
if (recordTrack->getCblk() == NULL) {
@@ -2432,7 +2411,9 @@ bool AudioFlinger::AudioRecordThread::threadLoop()
LOGV("AudioRecordThread: loop starting");
if (mRecordTrack != 0) {
- input = mAudioHardware->openInputStream(mRecordTrack->format(),
+ input = mAudioHardware->openInputStream(
+ mRecordTrack->inputSource(),
+ mRecordTrack->format(),
mRecordTrack->channelCount(),
mRecordTrack->sampleRate(),
&mStartStatus,
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index db5cc74..8e47b29 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -139,7 +139,7 @@ public:
// record interface
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
@@ -232,7 +232,6 @@ private:
TrackBase(const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
uint32_t sampleRate,
int format,
int channelCount,
@@ -260,10 +259,6 @@ private:
return mCblk;
}
- int type() const {
- return mStreamType;
- }
-
int format() const {
return mFormat;
}
@@ -293,7 +288,6 @@ private:
sp<Client> mClient;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
- int mStreamType;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
@@ -328,6 +322,11 @@ private:
void mute(bool);
void setVolume(float left, float right);
+ int type() const {
+ return mStreamType;
+ }
+
+
protected:
friend class MixerThread;
friend class AudioFlinger;
@@ -364,6 +363,7 @@ private:
int8_t mRetryCount;
sp<IMemory> mSharedBuffer;
bool mResetDone;
+ int mStreamType;
}; // end of Track
// record track
@@ -371,7 +371,7 @@ private:
public:
RecordTrack(const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
@@ -385,6 +385,8 @@ private:
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
+ int inputSource() const { return mInputSource; }
+
private:
friend class AudioFlinger;
friend class AudioFlinger::RecordHandle;
@@ -397,6 +399,7 @@ private:
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
bool mOverflow;
+ int mInputSource;
};
// playback track
@@ -501,7 +504,6 @@ private:
MixerThread& operator = (const MixerThread&);
status_t addTrack_l(const sp<Track>& track);
- void removeTrack_l(wp<Track> track, int name);
void destroyTrack_l(const sp<Track>& track);
int getTrackName_l();
void deleteTrackName_l(int name);
diff --git a/libs/audioflinger/AudioHardwareGeneric.cpp b/libs/audioflinger/AudioHardwareGeneric.cpp
index 62beada..1e159b8 100644
--- a/libs/audioflinger/AudioHardwareGeneric.cpp
+++ b/libs/audioflinger/AudioHardwareGeneric.cpp
@@ -30,6 +30,7 @@
#include <utils/String8.h>
#include "AudioHardwareGeneric.h"
+#include <media/AudioRecord.h>
namespace android {
@@ -93,9 +94,15 @@ void AudioHardwareGeneric::closeOutputStream(AudioStreamOutGeneric* out) {
}
AudioStreamIn* AudioHardwareGeneric::openInputStream(
- int format, int channelCount, uint32_t sampleRate, status_t *status,
- AudioSystem::audio_in_acoustics acoustics)
+ int inputSource, int format, int channelCount, uint32_t sampleRate,
+ status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
+ // check for valid input source
+ if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
+ (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+ return 0;
+ }
+
AutoMutex lock(mLock);
// only one input stream allowed
diff --git a/libs/audioflinger/AudioHardwareGeneric.h b/libs/audioflinger/AudioHardwareGeneric.h
index c949aa1..c89df87 100644
--- a/libs/audioflinger/AudioHardwareGeneric.h
+++ b/libs/audioflinger/AudioHardwareGeneric.h
@@ -112,6 +112,7 @@ public:
status_t *status=0);
virtual AudioStreamIn* openInputStream(
+ int inputSource,
int format,
int channelCount,
uint32_t sampleRate,
diff --git a/libs/audioflinger/AudioHardwareInterface.cpp b/libs/audioflinger/AudioHardwareInterface.cpp
index ac76a19..cc1bd8f 100644
--- a/libs/audioflinger/AudioHardwareInterface.cpp
+++ b/libs/audioflinger/AudioHardwareInterface.cpp
@@ -53,7 +53,7 @@ static const char* routeStrings[] =
"EARPIECE ",
"SPEAKER ",
"BLUETOOTH ",
- "HEADSET "
+ "HEADSET ",
"BLUETOOTH_A2DP "
};
static const char* routeNone = "NONE";
diff --git a/libs/audioflinger/AudioHardwareStub.cpp b/libs/audioflinger/AudioHardwareStub.cpp
index b13cb1c..0ab4c60 100644
--- a/libs/audioflinger/AudioHardwareStub.cpp
+++ b/libs/audioflinger/AudioHardwareStub.cpp
@@ -23,6 +23,7 @@
#include <utils/String8.h>
#include "AudioHardwareStub.h"
+#include <media/AudioRecord.h>
namespace android {
@@ -56,9 +57,15 @@ AudioStreamOut* AudioHardwareStub::openOutputStream(
}
AudioStreamIn* AudioHardwareStub::openInputStream(
- int format, int channelCount, uint32_t sampleRate,
+ int inputSource, int format, int channelCount, uint32_t sampleRate,
status_t *status, AudioSystem::audio_in_acoustics acoustics)
{
+ // check for valid input source
+ if ((inputSource < AudioRecord::DEFAULT_INPUT) ||
+ (inputSource >= AudioRecord::NUM_INPUT_SOURCES)) {
+ return 0;
+ }
+
AudioStreamInStub* in = new AudioStreamInStub();
status_t lStatus = in->set(format, channelCount, sampleRate, acoustics);
if (status) {
diff --git a/libs/audioflinger/AudioHardwareStub.h b/libs/audioflinger/AudioHardwareStub.h
index d406424..bf63cc5 100644
--- a/libs/audioflinger/AudioHardwareStub.h
+++ b/libs/audioflinger/AudioHardwareStub.h
@@ -78,6 +78,7 @@ public:
status_t *status=0);
virtual AudioStreamIn* openInputStream(
+ int inputSource,
int format,
int channelCount,
uint32_t sampleRate,
generated by cgit v1.1 at 2025-08-16 22:16:45 +0000