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-rw-r--r--cmds/stagefright/codec.cpp57
-rw-r--r--drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c9
-rw-r--r--include/media/AudioRecord.h8
-rw-r--r--include/media/AudioTrack.h12
-rw-r--r--include/media/IAudioFlinger.h3
-rw-r--r--media/libmedia/AudioRecord.cpp4
-rw-r--r--media/libmedia/AudioTrack.cpp17
-rw-r--r--media/libmedia/IAudioFlinger.cpp6
-rw-r--r--media/libmedia/JetPlayer.cpp2
-rw-r--r--media/libmedia/ToneGenerator.cpp2
-rw-r--r--media/libmediaplayerservice/MediaPlayerService.cpp4
-rw-r--r--media/libstagefright/AudioPlayer.cpp11
-rw-r--r--media/libstagefright/AudioSource.cpp5
-rw-r--r--media/libstagefright/codecs/aacenc/src/adj_thr.c4
-rw-r--r--media/libstagefright/codecs/amrwbenc/inc/basic_op.h4
-rw-r--r--media/libstagefright/foundation/AMessage.cpp6
-rw-r--r--services/audioflinger/AudioFlinger.cpp10
-rw-r--r--services/audioflinger/AudioFlinger.h11
18 files changed, 103 insertions, 72 deletions
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index 0414b98..fea62cc 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -49,6 +49,7 @@ struct CodecState {
size_t mCSDIndex;
Vector<sp<ABuffer> > mInBuffers;
Vector<sp<ABuffer> > mOutBuffers;
+ bool mSignalledInputEOS;
bool mSawOutputEOS;
int64_t mNumBuffersDecoded;
int64_t mNumBytesDecoded;
@@ -127,6 +128,7 @@ static int decode(
}
state->mCSDIndex = 0;
+ state->mSignalledInputEOS = false;
state->mSawOutputEOS = false;
ALOGV("got %d pieces of codec specific data.", state->mCSD.size());
@@ -180,33 +182,7 @@ static int decode(
status_t err = extractor->getSampleTrackIndex(&trackIndex);
if (err != OK) {
- ALOGV("signalling EOS.");
-
- for (size_t i = 0; i < stateByTrack.size(); ++i) {
- CodecState *state = &stateByTrack.editValueAt(i);
-
- for (;;) {
- size_t index;
- err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
-
- if (err == -EAGAIN) {
- continue;
- }
-
- CHECK_EQ(err, (status_t)OK);
-
- err = state->mCodec->queueInputBuffer(
- index,
- 0 /* offset */,
- 0 /* size */,
- 0ll /* timeUs */,
- MediaCodec::BUFFER_FLAG_EOS);
-
- CHECK_EQ(err, (status_t)OK);
- break;
- }
- }
-
+ ALOGV("saw input eos");
sawInputEOS = true;
} else {
CodecState *state = &stateByTrack.editValueFor(trackIndex);
@@ -240,6 +216,33 @@ static int decode(
CHECK_EQ(err, -EAGAIN);
}
}
+ } else {
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (!state->mSignalledInputEOS) {
+ size_t index;
+ status_t err =
+ state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("signalling input EOS on track %d", i);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ 0 /* size */,
+ 0ll /* timeUs */,
+ MediaCodec::BUFFER_FLAG_EOS);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ state->mSignalledInputEOS = true;
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
}
bool sawOutputEOSOnAllTracks = true;
diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
index dacf00e..365bdec 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
@@ -14,6 +14,7 @@
* limitations under the License.
*/
+#include <utils/Log.h>
#include <assert.h>
#include <errno.h>
#include <fcntl.h>
@@ -107,6 +108,7 @@ static int FwdLockFile_AcquireSession(int fileDesc) {
}
pthread_mutex_unlock(&sessionAcquisitionMutex);
if (i == MAX_NUM_SESSIONS) {
+ ALOGE("Too many sessions opened at the same time");
errno = ENFILE;
}
}
@@ -293,7 +295,12 @@ int FwdLockFile_attach(int fileDesc) {
int FwdLockFile_open(const char *pFilename) {
int fileDesc = open(pFilename, O_RDONLY);
- if (fileDesc >= 0 && FwdLockFile_attach(fileDesc) < 0) {
+ if (fileDesc < 0) {
+ ALOGE("failed to open file '%s': %s", pFilename, strerror(errno));
+ return fileDesc;
+ }
+
+ if (FwdLockFile_attach(fileDesc) < 0) {
(void)close(fileDesc);
fileDesc = -1;
}
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 4fbeb38..7df6668 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -143,6 +143,8 @@ public:
* user Context for use by the callback receiver.
*/
+ // FIXME consider removing this alias and replacing it by audio_in_acoustics_t
+ // or removing the parameter entirely if it is unused
enum record_flags {
RECORD_AGC_ENABLE = AUDIO_IN_ACOUSTICS_AGC_ENABLE,
RECORD_NS_ENABLE = AUDIO_IN_ACOUSTICS_NS_ENABLE,
@@ -154,7 +156,7 @@ public:
audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
- uint32_t flags = 0,
+ record_flags flags = (record_flags) 0,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
@@ -180,7 +182,7 @@ public:
audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
- uint32_t flags = 0,
+ record_flags flags = (record_flags) 0,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
@@ -383,7 +385,7 @@ private:
bool mMarkerReached;
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
- uint32_t mFlags;
+ record_flags mFlags;
uint32_t mChannelMask;
audio_io_handle_t mInput;
int mSessionId;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 95b9d86..552e829 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -153,7 +153,7 @@ public:
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
- uint32_t flags = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
@@ -165,7 +165,7 @@ public:
int format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
- uint32_t flags = 0,
+ uint32_t flags = (uint32_t) AUDIO_POLICY_OUTPUT_FLAG_NONE,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
@@ -185,7 +185,7 @@ public:
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
- uint32_t flags = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
@@ -209,7 +209,7 @@ public:
audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
- uint32_t flags = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
callback_t cbf = NULL,
void* user = NULL,
int notificationFrames = 0,
@@ -473,7 +473,7 @@ protected:
audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount);
@@ -515,7 +515,7 @@ protected:
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
- uint32_t mFlags;
+ audio_policy_output_flags_t mFlags;
int mSessionId;
int mAuxEffectId;
mutable Mutex mLock;
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 7a2ada0..c723b6d 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -28,6 +28,7 @@
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
#include <system/audio.h>
+#include <system/audio_policy.h>
#include <hardware/audio_effect.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
@@ -121,7 +122,7 @@ public:
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags) = 0;
+ audio_policy_output_flags_t flags) = 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/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 943f3af..22c3a18 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -89,7 +89,7 @@ AudioRecord::AudioRecord(
audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -124,7 +124,7 @@ status_t AudioRecord::set(
audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index a1c99e5..34563ca 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -92,7 +92,7 @@ AudioTrack::AudioTrack(
audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -119,10 +119,11 @@ AudioTrack::AudioTrack(
int notificationFrames,
int sessionId)
: mStatus(NO_INIT),
+ mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
- frameCount, flags, cbf, user, notificationFrames,
+ frameCount, (audio_policy_output_flags_t)flags, cbf, user, notificationFrames,
0, false, sessionId);
}
@@ -132,7 +133,7 @@ AudioTrack::AudioTrack(
audio_format_t format,
int channelMask,
const sp<IMemory>& sharedBuffer,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -172,7 +173,7 @@ status_t AudioTrack::set(
audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -221,7 +222,7 @@ status_t AudioTrack::set(
// force direct flag if format is not linear PCM
if (!audio_is_linear_pcm(format)) {
- flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
+ flags = (audio_policy_output_flags_t) (flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT);
}
if (!audio_is_output_channel(channelMask)) {
@@ -233,7 +234,7 @@ status_t AudioTrack::set(
audio_io_handle_t output = AudioSystem::getOutput(
streamType,
sampleRate, format, channelMask,
- (audio_policy_output_flags_t)flags);
+ flags);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
@@ -707,7 +708,7 @@ audio_io_handle_t AudioTrack::getOutput()
audio_io_handle_t AudioTrack::getOutput_l()
{
return AudioSystem::getOutput(mStreamType,
- mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
+ mCblk->sampleRate, mFormat, mChannelMask, mFlags);
}
int AudioTrack::getSessionId() const
@@ -734,7 +735,7 @@ status_t AudioTrack::createTrack_l(
audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount)
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index ebadbfa..47c261d 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -362,7 +362,7 @@ public:
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags)
+ audio_policy_output_flags_t flags)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
@@ -377,7 +377,7 @@ public:
data.writeInt32(format);
data.writeInt32(channels);
data.writeInt32(latency);
- data.writeInt32(flags);
+ data.writeInt32((int32_t) flags);
remote()->transact(OPEN_OUTPUT, data, &reply);
audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
ALOGV("openOutput() returned output, %d", output);
@@ -845,7 +845,7 @@ status_t BnAudioFlinger::onTransact(
audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
- uint32_t flags = data.readInt32();
+ audio_policy_output_flags_t flags = (audio_policy_output_flags_t) data.readInt32();
audio_io_handle_t output = openOutput(&devices,
&samplingRate,
&format,
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 6cb5b82..a85956c 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -94,7 +94,7 @@ int JetPlayer::init()
AUDIO_FORMAT_PCM_16_BIT,
(pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
mTrackBufferSize,
- 0);
+ AUDIO_POLICY_OUTPUT_FLAG_NONE);
// create render and playback thread
{
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 54eb98a..9c3170c 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -1024,7 +1024,7 @@ bool ToneGenerator::initAudioTrack() {
AUDIO_FORMAT_PCM_16_BIT,
AUDIO_CHANNEL_OUT_MONO,
0,
- 0,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
audioCallback,
this,
0,
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 1a85c9c..bbc53f3 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1503,7 +1503,7 @@ status_t MediaPlayerService::AudioOutput::open(
format,
channelMask,
frameCount,
- 0 /* flags */,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
CallbackWrapper,
this,
0,
@@ -1515,7 +1515,7 @@ status_t MediaPlayerService::AudioOutput::open(
format,
channelMask,
frameCount,
- 0,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
NULL,
NULL,
0,
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 650b6c4..23c3c74 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -151,7 +151,7 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) {
mAudioTrack = new AudioTrack(
AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
- 0, 0, &AudioCallback, this, 0);
+ 0, AUDIO_POLICY_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
delete mAudioTrack;
@@ -437,8 +437,11 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) {
kKeyTime, &mPositionTimeMediaUs));
mPositionTimeRealUs =
- ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
+ -mLatencyUs + ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
/ mSampleRate;
+ if (mPositionTimeRealUs < 0) {
+ mPositionTimeRealUs = 0;
+ }
ALOGV("buffer->size() = %d, "
"mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
@@ -493,7 +496,9 @@ int64_t AudioPlayer::getRealTimeUs() {
int64_t AudioPlayer::getRealTimeUsLocked() const {
CHECK(mStarted);
CHECK_NE(mSampleRate, 0);
- return -mLatencyUs + (mNumFramesPlayed * 1000000) / mSampleRate;
+ int64_t t = -mLatencyUs + (mNumFramesPlayed * 1000000) / mSampleRate;
+ if (t < 0) return 0;
+ return t;
}
int64_t AudioPlayer::getMediaTimeUs() {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 5b2ea1f..cbe709b 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -56,9 +56,10 @@ AudioSource::AudioSource(
ALOGV("sampleRate: %d, channels: %d", sampleRate, channels);
CHECK(channels == 1 || channels == 2);
- uint32_t flags = AudioRecord::RECORD_AGC_ENABLE |
+ AudioRecord::record_flags flags = (AudioRecord::record_flags)
+ (AudioRecord::RECORD_AGC_ENABLE |
AudioRecord::RECORD_NS_ENABLE |
- AudioRecord::RECORD_IIR_ENABLE;
+ AudioRecord::RECORD_IIR_ENABLE);
mRecord = new AudioRecord(
inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO,
diff --git a/media/libstagefright/codecs/aacenc/src/adj_thr.c b/media/libstagefright/codecs/aacenc/src/adj_thr.c
index 373b063..07b33b7 100644
--- a/media/libstagefright/codecs/aacenc/src/adj_thr.c
+++ b/media/libstagefright/codecs/aacenc/src/adj_thr.c
@@ -438,7 +438,7 @@ static void correctThresh(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS],
for (sfb=0; sfb<psyOutChan->maxSfbPerGroup; sfb++) {
Word32 redThrExp = thrExp[ch][sfbGrp+sfb] + redVal;
- if (((*pahFlag < AH_ACTIVE) || (deltaPe > 0)) && (redThrExp > 0) ) {
+ if (((*pahFlag < AH_ACTIVE) || (deltaPe > 0)) && (redThrExp > 0) && (redThrExp >= *psfbNActiveLines)) {
*psfbPeFactors = (*psfbNActiveLines) * (0x7fffffff / redThrExp);
normFactor = L_add(normFactor, *psfbPeFactors);
@@ -466,7 +466,7 @@ static void correctThresh(PSY_OUT_CHANNEL psyOutChannel[MAX_CHANNELS],
deltaSfbPe = *psfbPeFactors * deltaPe;
/* thr3(n) = thr2(n)*2^deltaSfbPe/b(n) */
- if (*psfbNActiveLines > 0) {
+ if (*psfbNActiveLines > 0 && (normFactor* (*psfbNActiveLines)) != 0) {
/* new threshold */
Word32 thrFactor;
sfbEn = psyOutChan->sfbEnergy[sfbGrp+sfb];
diff --git a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
index c23dce6..f42a27c 100644
--- a/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
+++ b/media/libstagefright/codecs/amrwbenc/inc/basic_op.h
@@ -29,11 +29,7 @@
#define MIN_16 (Word16)-32768 /* 0x8000 */
-#ifdef LINUX
-#define static_vo static __inline__
-#else
#define static_vo static __inline
-#endif
#define saturate(L_var1) (((L_var1) > 0X00007fffL) ? (MAX_16): (((L_var1) < (Word32) 0xffff8000L) ? (MIN_16): ((L_var1) & 0xffff)))
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index 9a00186..8b01ac6 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -74,6 +74,7 @@ void AMessage::freeItem(Item *item) {
case kTypeObject:
case kTypeMessage:
+ case kTypeBuffer:
{
if (item->u.refValue != NULL) {
item->u.refValue->decStrong(this);
@@ -292,6 +293,7 @@ sp<AMessage> AMessage::dup() const {
}
case kTypeObject:
+ case kTypeBuffer:
{
to->u.refValue = from->u.refValue;
to->u.refValue->incStrong(msg.get());
@@ -396,6 +398,10 @@ AString AMessage::debugString(int32_t indent) const {
tmp = StringPrintf(
"RefBase *%s = %p", item.mName, item.u.refValue);
break;
+ case kTypeBuffer:
+ tmp = StringPrintf(
+ "ABuffer *%s = %p", item.mName, item.u.refValue);
+ break;
case kTypeMessage:
tmp = StringPrintf(
"AMessage %s = %s",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f9b033b..f94ab84 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -5192,8 +5192,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
- (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
- (reqChannelCount < 3)) {
+ popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
+ (reqChannelCount <= FCC_2)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
@@ -5270,7 +5270,7 @@ void AudioFlinger::RecordThread::readInputParameters()
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
- if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+ if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
{
int channelCount;
// optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
@@ -5356,7 +5356,7 @@ audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices,
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags)
+ audio_policy_output_flags_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
@@ -5559,7 +5559,7 @@ audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices,
if (inStream == NULL && status == BAD_VALUE &&
reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
- (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
+ (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
ALOGV("openInput() reopening with proposed sampling rate and channels");
status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2e259c0..38fff8c 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -58,6 +58,15 @@ class AudioResampler;
// ----------------------------------------------------------------------------
+// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
+// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
+// Adding full support for > 2 channel capture or playback would require more than simply changing
+// this #define. There is an independent hard-coded upper limit in AudioMixer;
+// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
+// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
+// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
+#define FCC_2 2 // FCC_2 = Fixed Channel Count 2
+
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
@@ -134,7 +143,7 @@ public:
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags);
+ audio_policy_output_flags_t flags);
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2);
generated by cgit v1.1 at 2024-05-17 11:08:38 +0000