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-rw-r--r--media/libmedia/AudioRecord.cpp48
-rw-r--r--media/libmedia/AudioSystem.cpp15
-rw-r--r--media/libmedia/IAudioFlinger.cpp25
-rw-r--r--media/libmedia/IAudioPolicyService.cpp5
4 files changed, 39 insertions, 54 deletions
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0562f8e..f8813c9 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,28 +18,18 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <sched.h>
#include <sys/resource.h>
+#include <sys/types.h>
-#include <private/media/AudioTrackShared.h>
-
-#include <media/AudioSystem.h>
+#include <binder/IPCThreadState.h>
+#include <cutils/atomic.h>
+#include <cutils/compiler.h>
#include <media/AudioRecord.h>
-#include <media/mediarecorder.h>
-
-#include <binder/IServiceManager.h>
+#include <media/AudioSystem.h>
+#include <system/audio.h>
#include <utils/Log.h>
-#include <binder/Parcel.h>
-#include <binder/IPCThreadState.h>
-#include <utils/Timers.h>
-#include <utils/Atomic.h>
-#include <system/audio.h>
-#include <cutils/bitops.h>
-#include <cutils/compiler.h>
+#include <private/media/AudioTrackShared.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -49,18 +39,18 @@ status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
audio_format_t format,
- int channelCount)
+ audio_channel_mask_t channelMask)
{
size_t size = 0;
- if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
+ if (AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size)
!= NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size.");
return NO_INIT;
}
if (size == 0) {
- ALOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
- sampleRate, format, channelCount);
+ ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+ sampleRate, format, channelMask);
return BAD_VALUE;
}
@@ -68,6 +58,7 @@ status_t AudioRecord::getMinFrameCount(
size <<= 1;
if (audio_is_linear_pcm(format)) {
+ int channelCount = popcount(channelMask);
size /= channelCount * audio_bytes_per_sample(format);
}
@@ -87,9 +78,8 @@ AudioRecord::AudioRecord(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
- record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -98,7 +88,7 @@ AudioRecord::AudioRecord(
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT)
{
mStatus = set(inputSource, sampleRate, format, channelMask,
- frameCount, flags, cbf, user, notificationFrames, sessionId);
+ frameCount, cbf, user, notificationFrames, sessionId);
}
AudioRecord::~AudioRecord()
@@ -122,9 +112,8 @@ status_t AudioRecord::set(
audio_source_t inputSource,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
- record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -132,7 +121,7 @@ status_t AudioRecord::set(
int sessionId)
{
- ALOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount);
+ ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount);
AutoMutex lock(mLock);
@@ -174,7 +163,6 @@ status_t AudioRecord::set(
sampleRate,
format,
channelMask,
- (audio_in_acoustics_t)flags,
mSessionId);
if (input == 0) {
ALOGE("Could not get audio input for record source %d", inputSource);
@@ -229,7 +217,6 @@ status_t AudioRecord::set(
mNewPosition = 0;
mUpdatePeriod = 0;
mInputSource = inputSource;
- mFlags = flags;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
@@ -457,7 +444,7 @@ unsigned int AudioRecord::getInputFramesLost() const
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
audio_io_handle_t input)
{
@@ -613,7 +600,6 @@ audio_io_handle_t AudioRecord::getInput_l()
mCblk->sampleRate,
mFormat,
mChannelMask,
- (audio_in_acoustics_t)mFlags,
mSessionId);
return mInput;
}
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 4c41ba5..9c270c8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -41,7 +41,7 @@ DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSyst
// Cached values for recording queries, all protected by gLock
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-int AudioSystem::gPrevInChannelCount = 1;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
size_t AudioSystem::gInBuffSize = 0;
@@ -334,25 +334,25 @@ status_t AudioSystem::getLatency(audio_io_handle_t output,
return NO_ERROR;
}
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
- size_t* buffSize)
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask, size_t* buffSize)
{
gLock.lock();
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
size_t inBuffSize = gInBuffSize;
if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
- || (channelCount != gPrevInChannelCount)) {
+ || (channelMask != gPrevInChannelMask)) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) {
return PERMISSION_DENIED;
}
- inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
- gPrevInChannelCount = channelCount;
+ gPrevInChannelMask = channelMask;
gInBuffSize = inBuffSize;
}
@@ -625,12 +625,11 @@ audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
uint32_t channels,
- audio_in_acoustics_t acoustics,
int sessionId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getInput(inputSource, samplingRate, format, channels, acoustics, sessionId);
+ return aps->getInput(inputSource, samplingRate, format, channels, sessionId);
}
status_t AudioSystem::startInput(audio_io_handle_t input)
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index e8dd438..27f6b45 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -32,7 +32,7 @@ enum {
CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
OPEN_RECORD,
SAMPLE_RATE,
- CHANNEL_COUNT,
+ CHANNEL_COUNT, // obsolete
FORMAT,
FRAME_COUNT,
LATENCY,
@@ -86,7 +86,7 @@ public:
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
track_flags_t flags,
const sp<IMemory>& sharedBuffer,
@@ -182,6 +182,7 @@ public:
return reply.readInt32();
}
+#if 0
virtual int channelCount(audio_io_handle_t output) const
{
Parcel data, reply;
@@ -190,6 +191,7 @@ public:
remote()->transact(CHANNEL_COUNT, data, &reply);
return reply.readInt32();
}
+#endif
virtual audio_format_t format(audio_io_handle_t output) const
{
@@ -347,13 +349,14 @@ public:
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+ audio_channel_mask_t channelMask) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(sampleRate);
data.writeInt32(format);
- data.writeInt32(channelCount);
+ data.writeInt32(channelMask);
remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
return reply.readInt32();
}
@@ -698,7 +701,7 @@ status_t BnAudioFlinger::onTransact(
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
size_t bufferCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
@@ -708,7 +711,7 @@ status_t BnAudioFlinger::onTransact(
status_t status;
sp<IAudioTrack> track = createTrack(pid,
(audio_stream_type_t) streamType, sampleRate, format,
- channelCount, bufferCount, flags, buffer, output, tid, &sessionId, &status);
+ channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(track->asBinder());
@@ -720,13 +723,13 @@ status_t BnAudioFlinger::onTransact(
audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
size_t bufferCount = data.readInt32();
track_flags_t flags = (track_flags_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(pid, input,
- sampleRate, format, channelCount, bufferCount, flags, &sessionId, &status);
+ sampleRate, format, channelMask, bufferCount, flags, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(record->asBinder());
@@ -737,11 +740,13 @@ status_t BnAudioFlinger::onTransact(
reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
+#if 0
case CHANNEL_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
+#endif
case FORMAT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
@@ -846,8 +851,8 @@ status_t BnAudioFlinger::onTransact(
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- int channelCount = data.readInt32();
- reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
+ audio_channel_mask_t channelMask = data.readInt32();
+ reply->writeInt32( getInputBufferSize(sampleRate, format, channelMask) );
return NO_ERROR;
} break;
case OPEN_OUTPUT: {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 7aab8d6..5a4512e 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -176,7 +176,6 @@ public:
uint32_t samplingRate,
audio_format_t format,
uint32_t channels,
- audio_in_acoustics_t acoustics,
int audioSession)
{
Parcel data, reply;
@@ -185,7 +184,6 @@ public:
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channels);
- data.writeInt32(static_cast <uint32_t>(acoustics));
data.writeInt32(audioSession);
remote()->transact(GET_INPUT, data, &reply);
return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -465,14 +463,11 @@ status_t BnAudioPolicyService::onTransact(
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
- audio_in_acoustics_t acoustics =
- static_cast <audio_in_acoustics_t>(data.readInt32());
int audioSession = data.readInt32();
audio_io_handle_t input = getInput(inputSource,
samplingRate,
format,
channels,
- acoustics,
audioSession);
reply->writeInt32(static_cast <int>(input));
return NO_ERROR;
generated by cgit v1.1 at 2025-11-27 10:01:51 +0000