audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 5d74a0a..8438714 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -37,6 +37,9 @@
#include <utils/Timers.h>
#include <utils/Atomic.h>
+#include <hardware/audio.h>
+#include <cutils/bitops.h>
+
#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
@@ -66,8 +69,8 @@
// We double the size of input buffer for ping pong use of record buffer.
size <<= 1;
- if (AudioSystem::isLinearPCM(format)) {
- size /= channelCount * (format == AudioSystem::PCM_16_BIT ? 2 : 1);
+ if (audio_is_linear_pcm(format)) {
+ size /= channelCount * (format == AUDIO_FORMAT_PCM_16_BIT ? 2 : 1);
}
*frameCount = size;
@@ -145,22 +148,22 @@
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
- format = AudioSystem::PCM_16_BIT;
+ format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
- if (!AudioSystem::isValidFormat(format)) {
+ if (!audio_is_valid_format(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
- if (!AudioSystem::isInputChannel(channels)) {
+ if (!audio_is_input_channel(channels)) {
return BAD_VALUE;
}
- int channelCount = AudioSystem::popCount(channels);
+ int channelCount = popcount(channels);
audio_io_handle_t input = AudioSystem::getInput(inputSource,
- sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
+ sampleRate, format, channels, (audio_in_acoustics_t)flags);
if (input == 0) {
LOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
@@ -254,8 +257,8 @@
int AudioRecord::frameSize() const
{
- if (AudioSystem::isLinearPCM(mFormat)) {
- return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+ if (audio_is_linear_pcm(mFormat)) {
+ return channelCount()*((format() == AUDIO_FORMAT_PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
} else {
return sizeof(uint8_t);
}
@@ -587,7 +590,7 @@
mInput = AudioSystem::getInput(mInputSource,
mCblk->sampleRate,
mFormat, mChannels,
- (AudioSystem::audio_in_acoustics)mFlags);
+ (audio_in_acoustics_t)mFlags);
return mInput;
}
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 5c6f344..e08a55b 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -23,6 +23,8 @@
#include <media/IAudioPolicyService.h>
#include <math.h>
+#include <hardware/audio.h>
+
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
@@ -45,7 +47,7 @@
// Cached values for recording queries
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT;
+int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
int AudioSystem::gPrevInChannelCount = 1;
size_t AudioSystem::gInBuffSize = 0;
@@ -127,7 +129,7 @@
status_t AudioSystem::setStreamVolume(int stream, float value, int output)
{
- if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setStreamVolume(stream, value, output);
@@ -136,7 +138,7 @@
status_t AudioSystem::setStreamMute(int stream, bool mute)
{
- if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setStreamMute(stream, mute);
@@ -145,7 +147,7 @@
status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
{
- if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*volume = af->streamVolume(stream, output);
@@ -154,7 +156,7 @@
status_t AudioSystem::getStreamMute(int stream, bool* mute)
{
- if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*mute = af->streamMute(stream);
@@ -163,7 +165,7 @@
status_t AudioSystem::setMode(int mode)
{
- if (mode >= NUM_MODES) return BAD_VALUE;
+ if (mode >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMode(mode);
@@ -213,11 +215,11 @@
OutputDescriptor *outputDesc;
audio_io_handle_t output;
- if (streamType == DEFAULT) {
- streamType = MUSIC;
+ if (streamType == AUDIO_STREAM_DEFAULT) {
+ streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((stream_type)streamType);
+ output = getOutput((audio_stream_type_t)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
@@ -246,11 +248,11 @@
OutputDescriptor *outputDesc;
audio_io_handle_t output;
- if (streamType == DEFAULT) {
- streamType = MUSIC;
+ if (streamType == AUDIO_STREAM_DEFAULT) {
+ streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((stream_type)streamType);
+ output = getOutput((audio_stream_type_t)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
@@ -277,11 +279,11 @@
OutputDescriptor *outputDesc;
audio_io_handle_t output;
- if (streamType == DEFAULT) {
- streamType = MUSIC;
+ if (streamType == AUDIO_STREAM_DEFAULT) {
+ streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((stream_type)streamType);
+ output = getOutput((audio_stream_type_t)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
@@ -338,11 +340,11 @@
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
- if (stream == DEFAULT) {
- stream = MUSIC;
+ if (stream == AUDIO_STREAM_DEFAULT) {
+ stream = AUDIO_STREAM_MUSIC;
}
- return af->getRenderPosition(halFrames, dspFrames, getOutput((stream_type)stream));
+ return af->getRenderPosition(halFrames, dspFrames, getOutput((audio_stream_type_t)stream));
}
unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -455,10 +457,10 @@
bool AudioSystem::routedToA2dpOutput(int streamType) {
switch(streamType) {
- case MUSIC:
- case VOICE_CALL:
- case BLUETOOTH_SCO:
- case SYSTEM:
+ case AUDIO_STREAM_MUSIC:
+ case AUDIO_STREAM_VOICE_CALL:
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ case AUDIO_STREAM_SYSTEM:
return true;
default:
return false;
@@ -497,9 +499,9 @@
return gAudioPolicyService;
}
-status_t AudioSystem::setDeviceConnectionState(audio_devices device,
- device_connection_state state,
- const char *device_address)
+status_t AudioSystem::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
@@ -507,11 +509,11 @@
return aps->setDeviceConnectionState(device, state, device_address);
}
-AudioSystem::device_connection_state AudioSystem::getDeviceConnectionState(audio_devices device,
+audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return DEVICE_STATE_UNAVAILABLE;
+ if (aps == 0) return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
return aps->getDeviceConnectionState(device, device_address);
}
@@ -531,26 +533,26 @@
return aps->setRingerMode(mode, mask);
}
-status_t AudioSystem::setForceUse(force_use usage, forced_config config)
+status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setForceUse(usage, config);
}
-AudioSystem::forced_config AudioSystem::getForceUse(force_use usage)
+audio_policy_forced_cfg_t AudioSystem::getForceUse(audio_policy_force_use_t usage)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return FORCE_NONE;
+ if (aps == 0) return AUDIO_POLICY_FORCE_NONE;
return aps->getForceUse(usage);
}
-audio_io_handle_t AudioSystem::getOutput(stream_type stream,
+audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
- output_flags flags)
+ audio_policy_output_flags_t flags)
{
audio_io_handle_t output = 0;
// Do not use stream to output map cache if the direct output
@@ -561,9 +563,9 @@
// be reworked for proper operation with direct outputs. This code is too specific
// to the first use case we want to cover (Voice Recognition and Voice Dialer over
// Bluetooth SCO
- if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0 &&
- ((stream != AudioSystem::VOICE_CALL && stream != AudioSystem::BLUETOOTH_SCO) ||
- channels != AudioSystem::CHANNEL_OUT_MONO ||
+ if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) == 0 &&
+ ((stream != AUDIO_STREAM_VOICE_CALL && stream != AUDIO_STREAM_BLUETOOTH_SCO) ||
+ channels != AUDIO_CHANNEL_OUT_MONO ||
(samplingRate != 8000 && samplingRate != 16000))) {
Mutex::Autolock _l(gLock);
output = AudioSystem::gStreamOutputMap.valueFor(stream);
@@ -573,7 +575,7 @@
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
output = aps->getOutput(stream, samplingRate, format, channels, flags);
- if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
+ if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) == 0) {
Mutex::Autolock _l(gLock);
AudioSystem::gStreamOutputMap.add(stream, output);
}
@@ -582,7 +584,7 @@
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -591,7 +593,7 @@
}
status_t AudioSystem::stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -610,7 +612,7 @@
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
- audio_in_acoustics acoustics)
+ audio_in_acoustics_t acoustics)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
@@ -638,7 +640,7 @@
aps->releaseInput(input);
}
-status_t AudioSystem::initStreamVolume(stream_type stream,
+status_t AudioSystem::initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax)
{
@@ -647,28 +649,28 @@
return aps->initStreamVolume(stream, indexMin, indexMax);
}
-status_t AudioSystem::setStreamVolumeIndex(stream_type stream, int index)
+status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream, int index)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setStreamVolumeIndex(stream, index);
}
-status_t AudioSystem::getStreamVolumeIndex(stream_type stream, int *index)
+status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->getStreamVolumeIndex(stream, index);
}
-uint32_t AudioSystem::getStrategyForStream(AudioSystem::stream_type stream)
+uint32_t AudioSystem::getStrategyForStream(audio_stream_type_t stream)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
return aps->getStrategyForStream(stream);
}
-uint32_t AudioSystem::getDevicesForStream(AudioSystem::stream_type stream)
+uint32_t AudioSystem::getDevicesForStream(audio_stream_type_t stream)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
@@ -717,122 +719,5 @@
LOGW("AudioPolicyService server died!");
}
-// ---------------------------------------------------------------------------
-
-
-// use emulated popcount optimization
-// http://www.df.lth.se/~john_e/gems/gem002d.html
-uint32_t AudioSystem::popCount(uint32_t u)
-{
- u = ((u&0x55555555) + ((u>>1)&0x55555555));
- u = ((u&0x33333333) + ((u>>2)&0x33333333));
- u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
- u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
- u = ( u&0x0000ffff) + (u>>16);
- return u;
-}
-
-bool AudioSystem::isOutputDevice(audio_devices device)
-{
- if ((popCount(device) == 1 ) &&
- ((device & ~AudioSystem::DEVICE_OUT_ALL) == 0)) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isInputDevice(audio_devices device)
-{
- if ((popCount(device) == 1 ) &&
- ((device & ~AudioSystem::DEVICE_IN_ALL) == 0)) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isA2dpDevice(audio_devices device)
-{
- if ((popCount(device) == 1 ) &&
- (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
- AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER))) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isBluetoothScoDevice(audio_devices device)
-{
- if ((popCount(device) == 1 ) &&
- (device & (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO |
- AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
- AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT |
- AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET))) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isLowVisibility(stream_type stream)
-{
- if (stream == AudioSystem::SYSTEM ||
- stream == AudioSystem::NOTIFICATION ||
- stream == AudioSystem::RING) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isInputChannel(uint32_t channel)
-{
- if ((channel & ~AudioSystem::CHANNEL_IN_ALL) == 0) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isOutputChannel(uint32_t channel)
-{
- if ((channel & ~AudioSystem::CHANNEL_OUT_ALL) == 0) {
- return true;
- } else {
- return false;
- }
-}
-
-bool AudioSystem::isValidFormat(uint32_t format)
-{
- switch (format & MAIN_FORMAT_MASK) {
- case PCM:
- case MP3:
- case AMR_NB:
- case AMR_WB:
- case AAC:
- case HE_AAC_V1:
- case HE_AAC_V2:
- case VORBIS:
- return true;
- default:
- return false;
- }
-}
-
-bool AudioSystem::isLinearPCM(uint32_t format)
-{
- switch (format) {
- case PCM_16_BIT:
- case PCM_8_BIT:
- return true;
- default:
- return false;
- }
-}
-
}; // namespace android
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 66e11d2..2673df9 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -37,6 +37,11 @@
#include <utils/Timers.h>
#include <utils/Atomic.h>
+#include <cutils/bitops.h>
+
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+
#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
@@ -165,39 +170,41 @@
}
// handle default values first.
- if (streamType == AudioSystem::DEFAULT) {
- streamType = AudioSystem::MUSIC;
+ if (streamType == AUDIO_STREAM_DEFAULT) {
+ streamType = AUDIO_STREAM_MUSIC;
}
if (sampleRate == 0) {
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
- format = AudioSystem::PCM_16_BIT;
+ format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channels == 0) {
- channels = AudioSystem::CHANNEL_OUT_STEREO;
+ channels = AUDIO_CHANNEL_OUT_STEREO;
}
// validate parameters
- if (!AudioSystem::isValidFormat(format)) {
+ if (!audio_is_valid_format(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
- if (!AudioSystem::isLinearPCM(format)) {
- flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
+ if (!audio_is_linear_pcm(format)) {
+ flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
}
- if (!AudioSystem::isOutputChannel(channels)) {
+ if (!audio_is_output_channel(channels)) {
LOGE("Invalid channel mask");
return BAD_VALUE;
}
- uint32_t channelCount = AudioSystem::popCount(channels);
+ uint32_t channelCount = popcount(channels);
- audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
- sampleRate, format, channels, (AudioSystem::output_flags)flags);
+ audio_io_handle_t output = AudioSystem::getOutput(
+ (audio_stream_type_t)streamType,
+ sampleRate,format, channels,
+ (audio_policy_output_flags_t)flags);
if (output == 0) {
LOGE("Could not get audio output for stream type %d", streamType);
@@ -290,8 +297,8 @@
int AudioTrack::frameSize() const
{
- if (AudioSystem::isLinearPCM(mFormat)) {
- return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+ if (audio_is_linear_pcm(mFormat)) {
+ return channelCount()*((format() == AUDIO_FORMAT_PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
} else {
return sizeof(uint8_t);
}
@@ -673,8 +680,8 @@
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
- return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
- mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
+ return AudioSystem::getOutput((audio_stream_type_t)mStreamType,
+ mCblk->sampleRate, mFormat, mChannels, (audio_policy_output_flags_t)mFlags);
}
int AudioTrack::getSessionId()
@@ -727,7 +734,7 @@
}
mNotificationFramesAct = mNotificationFramesReq;
- if (!AudioSystem::isLinearPCM(format)) {
+ if (!audio_is_linear_pcm(format)) {
if (sharedBuffer != 0) {
frameCount = sharedBuffer->size();
}
@@ -923,8 +930,8 @@
audioBuffer->channelCount = mChannelCount;
audioBuffer->frameCount = framesReq;
audioBuffer->size = framesReq * cblk->frameSize;
- if (AudioSystem::isLinearPCM(mFormat)) {
- audioBuffer->format = AudioSystem::PCM_16_BIT;
+ if (audio_is_linear_pcm(mFormat)) {
+ audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT;
} else {
audioBuffer->format = mFormat;
}
@@ -982,7 +989,7 @@
size_t toWrite;
- if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
+ if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
// 8 to 16 bit conversion
@@ -1085,7 +1092,7 @@
// Divide buffer size by 2 to take into account the expansion
// due to 8 to 16 bit conversion: the callback must fill only half
// of the destination buffer
- if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
+ if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
@@ -1104,7 +1111,7 @@
}
if (writtenSize > reqSize) writtenSize = reqSize;
- if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
+ if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion
const int8_t *src = audioBuffer.i8 + writtenSize-1;
int count = writtenSize;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index b89a278..88a9ae0 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -25,6 +25,8 @@
#include <media/IAudioPolicyService.h>
+#include <hardware/audio.h>
+
namespace android {
enum {
@@ -62,8 +64,8 @@
}
virtual status_t setDeviceConnectionState(
- AudioSystem::audio_devices device,
- AudioSystem::device_connection_state state,
+ audio_devices_t device,
+ audio_policy_dev_state_t state,
const char *device_address)
{
Parcel data, reply;
@@ -75,8 +77,8 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual AudioSystem::device_connection_state getDeviceConnectionState(
- AudioSystem::audio_devices device,
+ virtual audio_policy_dev_state_t getDeviceConnectionState(
+ audio_devices_t device,
const char *device_address)
{
Parcel data, reply;
@@ -84,7 +86,7 @@
data.writeInt32(static_cast <uint32_t>(device));
data.writeCString(device_address);
remote()->transact(GET_DEVICE_CONNECTION_STATE, data, &reply);
- return static_cast <AudioSystem::device_connection_state>(reply.readInt32());
+ return static_cast <audio_policy_dev_state_t>(reply.readInt32());
}
virtual status_t setPhoneState(int state)
@@ -106,7 +108,7 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
+ virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -116,21 +118,21 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage)
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(usage));
remote()->transact(GET_FORCE_USE, data, &reply);
- return static_cast <AudioSystem::forced_config> (reply.readInt32());
+ return static_cast <audio_policy_forced_cfg_t> (reply.readInt32());
}
virtual audio_io_handle_t getOutput(
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
- AudioSystem::output_flags flags)
+ audio_policy_output_flags_t flags)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -144,7 +146,7 @@
}
virtual status_t startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session)
{
Parcel data, reply;
@@ -157,7 +159,7 @@
}
virtual status_t stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session)
{
Parcel data, reply;
@@ -182,7 +184,7 @@
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
- AudioSystem::audio_in_acoustics acoustics)
+ audio_in_acoustics_t acoustics)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -221,7 +223,7 @@
remote()->transact(RELEASE_INPUT, data, &reply);
}
- virtual status_t initStreamVolume(AudioSystem::stream_type stream,
+ virtual status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax)
{
@@ -234,7 +236,7 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -244,7 +246,7 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -255,7 +257,7 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream)
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -264,7 +266,7 @@
return reply.readInt32();
}
- virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream)
+ virtual uint32_t getDevicesForStream(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -330,10 +332,10 @@
switch(code) {
case SET_DEVICE_CONNECTION_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::audio_devices device =
- static_cast <AudioSystem::audio_devices>(data.readInt32());
- AudioSystem::device_connection_state state =
- static_cast <AudioSystem::device_connection_state>(data.readInt32());
+ audio_devices_t device =
+ static_cast <audio_devices_t>(data.readInt32());
+ audio_policy_dev_state_t state =
+ static_cast <audio_policy_dev_state_t>(data.readInt32());
const char *device_address = data.readCString();
reply->writeInt32(static_cast<uint32_t> (setDeviceConnectionState(device,
state,
@@ -343,8 +345,8 @@
case GET_DEVICE_CONNECTION_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::audio_devices device =
- static_cast<AudioSystem::audio_devices> (data.readInt32());
+ audio_devices_t device =
+ static_cast<audio_devices_t> (data.readInt32());
const char *device_address = data.readCString();
reply->writeInt32(static_cast<uint32_t> (getDeviceConnectionState(device,
device_address)));
@@ -367,29 +369,29 @@
case SET_FORCE_USE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::force_use usage = static_cast <AudioSystem::force_use>(data.readInt32());
- AudioSystem::forced_config config =
- static_cast <AudioSystem::forced_config>(data.readInt32());
+ audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32());
+ audio_policy_forced_cfg_t config =
+ static_cast <audio_policy_forced_cfg_t>(data.readInt32());
reply->writeInt32(static_cast <uint32_t>(setForceUse(usage, config)));
return NO_ERROR;
} break;
case GET_FORCE_USE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::force_use usage = static_cast <AudioSystem::force_use>(data.readInt32());
+ audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32());
reply->writeInt32(static_cast <uint32_t>(getForceUse(usage)));
return NO_ERROR;
} break;
case GET_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
uint32_t format = data.readInt32();
uint32_t channels = data.readInt32();
- AudioSystem::output_flags flags =
- static_cast <AudioSystem::output_flags>(data.readInt32());
+ audio_policy_output_flags_t flags =
+ static_cast <audio_policy_output_flags_t>(data.readInt32());
audio_io_handle_t output = getOutput(stream,
samplingRate,
@@ -406,7 +408,7 @@
uint32_t stream = data.readInt32();
int session = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(startOutput(output,
- (AudioSystem::stream_type)stream,
+ (audio_stream_type_t)stream,
session)));
return NO_ERROR;
} break;
@@ -417,7 +419,7 @@
uint32_t stream = data.readInt32();
int session = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
- (AudioSystem::stream_type)stream,
+ (audio_stream_type_t)stream,
session)));
return NO_ERROR;
} break;
@@ -435,8 +437,8 @@
uint32_t samplingRate = data.readInt32();
uint32_t format = data.readInt32();
uint32_t channels = data.readInt32();
- AudioSystem::audio_in_acoustics acoustics =
- static_cast <AudioSystem::audio_in_acoustics>(data.readInt32());
+ audio_in_acoustics_t acoustics =
+ static_cast <audio_in_acoustics_t>(data.readInt32());
audio_io_handle_t input = getInput(inputSource,
samplingRate,
format,
@@ -469,8 +471,8 @@
case INIT_STREAM_VOLUME: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
int indexMin = data.readInt32();
int indexMax = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(initStreamVolume(stream, indexMin,indexMax)));
@@ -479,8 +481,8 @@
case SET_STREAM_VOLUME: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
int index = data.readInt32();
reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream, index)));
return NO_ERROR;
@@ -488,8 +490,8 @@
case GET_STREAM_VOLUME: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
int index;
status_t status = getStreamVolumeIndex(stream, &index);
reply->writeInt32(index);
@@ -499,16 +501,16 @@
case GET_STRATEGY_FOR_STREAM: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
reply->writeInt32(getStrategyForStream(stream));
return NO_ERROR;
} break;
case GET_DEVICES_FOR_STREAM: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- AudioSystem::stream_type stream =
- static_cast <AudioSystem::stream_type>(data.readInt32());
+ audio_stream_type_t stream =
+ static_cast <audio_stream_type_t>(data.readInt32());
reply->writeInt32(static_cast <int>(getDevicesForStream(stream)));
return NO_ERROR;
} break;
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index ee9e1d8..88157d2 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -96,10 +96,10 @@
// create the output AudioTrack
mAudioTrack = new AudioTrack();
- mAudioTrack->set(AudioSystem::MUSIC, //TODO parametrize this
+ mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parametrize this
pLibConfig->sampleRate,
1, // format = PCM 16bits per sample,
- (pLibConfig->numChannels == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
+ (pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
mTrackBufferSize,
0);
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 82fe2d4..9f1b3d6 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -1026,8 +1026,8 @@
mpAudioTrack->set(mStreamType,
0,
- AudioSystem::PCM_16_BIT,
- AudioSystem::CHANNEL_OUT_MONO,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_OUT_MONO,
0,
0,
audioCallback,
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index 43571cf..366707c 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -24,6 +24,8 @@
#include <sys/types.h>
#include <limits.h>
+#include <cutils/bitops.h>
+
#include <media/Visualizer.h>
extern void fixed_fft_real(int n, int32_t *v);
@@ -127,7 +129,7 @@
{
if (size > VISUALIZER_CAPTURE_SIZE_MAX ||
size < VISUALIZER_CAPTURE_SIZE_MIN ||
- AudioSystem::popCount(size) != 1) {
+ popcount(size) != 1) {
return BAD_VALUE;
}
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index e80e742..9daa80f 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -37,6 +37,8 @@
#include <utils/KeyedVector.h>
#include <utils/String8.h>
+#include <hardware/audio.h>
+
namespace android {
MediaPlayer::MediaPlayer()
@@ -45,7 +47,7 @@
mListener = NULL;
mCookie = NULL;
mDuration = -1;
- mStreamType = AudioSystem::MUSIC;
+ mStreamType = AUDIO_STREAM_MUSIC;
mCurrentPosition = -1;
mSeekPosition = -1;
mCurrentState = MEDIA_PLAYER_IDLE;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 6b97708..9dd353b 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -53,6 +53,8 @@
#include <media/AudioTrack.h>
#include <media/MemoryLeakTrackUtil.h>
+#include <hardware/audio.h>
+
#include <private/android_filesystem_config.h>
#include "MediaRecorderClient.h"
@@ -1209,7 +1211,7 @@
mSessionId(sessionId) {
LOGV("AudioOutput(%d)", sessionId);
mTrack = 0;
- mStreamType = AudioSystem::MUSIC;
+ mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
mLatency = 0;
@@ -1319,7 +1321,7 @@
mStreamType,
sampleRate,
format,
- (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
+ (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
frameCount,
0 /* flags */,
CallbackWrapper,
@@ -1331,7 +1333,7 @@
mStreamType,
sampleRate,
format,
- (channelCount == 2) ? AudioSystem::CHANNEL_OUT_STEREO : AudioSystem::CHANNEL_OUT_MONO,
+ (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
frameCount,
0,
NULL,
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 5539a37..31b518e 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -30,6 +30,8 @@
#include <media/MediaPlayerInterface.h>
#include <media/Metadata.h>
+#include <hardware/audio.h>
+
namespace android {
class IMediaRecorder;
@@ -130,7 +132,7 @@
virtual ssize_t bufferSize() const { return frameSize() * mFrameCount; }
virtual ssize_t frameCount() const { return mFrameCount; }
virtual ssize_t channelCount() const { return (ssize_t)mChannelCount; }
- virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); }
+ virtual ssize_t frameSize() const { return ssize_t(mChannelCount * ((mFormat == AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(u_int8_t))); }
virtual uint32_t latency() const;
virtual float msecsPerFrame() const;
virtual status_t getPosition(uint32_t *position);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index 1a1780c..5a47384 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -35,6 +35,8 @@
#include <media/AudioTrack.h>
+#include <hardware/audio.h>
+
#include "MediaRecorderClient.h"
#include "MediaPlayerService.h"
@@ -102,7 +104,7 @@
LOGE("recorder is not initialized");
return NO_INIT;
}
- return mRecorder->setAudioSource((audio_source)as);
+ return mRecorder->setAudioSource((audio_source_t)as);
}
status_t MediaRecorderClient::setOutputFormat(int of)
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index 1b0b05f..37a3db3 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -30,6 +30,8 @@
#include <sys/types.h>
#include <sys/stat.h>
+#include <hardware/audio.h>
+
#include "MidiFile.h"
#ifdef HAVE_GETTID
@@ -58,7 +60,7 @@
MidiFile::MidiFile() :
mEasData(NULL), mEasHandle(NULL), mAudioBuffer(NULL),
mPlayTime(-1), mDuration(-1), mState(EAS_STATE_ERROR),
- mStreamType(AudioSystem::MUSIC), mLoop(false), mExit(false),
+ mStreamType(AUDIO_STREAM_MUSIC), mLoop(false), mExit(false),
mPaused(false), mRender(false), mTid(-1)
{
LOGV("constructor");
@@ -423,7 +425,7 @@
}
status_t MidiFile::createOutputTrack() {
- if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels, AudioSystem::PCM_16_BIT, 2) != NO_ERROR) {
+ if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels, AUDIO_FORMAT_PCM_16_BIT, 2) != NO_ERROR) {
LOGE("mAudioSink open failed");
return ERROR_OPEN_FAILED;
}
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index e3dfabb..01fbea1 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -46,6 +46,8 @@
#include <ctype.h>
#include <unistd.h>
+#include <hardware/audio.h>
+
#include "ARTPWriter.h"
namespace android {
@@ -64,7 +66,7 @@
StagefrightRecorder::StagefrightRecorder()
: mWriter(NULL), mWriterAux(NULL),
mOutputFd(-1), mOutputFdAux(-1),
- mAudioSource(AUDIO_SOURCE_LIST_END),
+ mAudioSource(AUDIO_SOURCE_CNT),
mVideoSource(VIDEO_SOURCE_LIST_END),
mStarted(false) {
@@ -82,10 +84,10 @@
return OK;
}
-status_t StagefrightRecorder::setAudioSource(audio_source as) {
+status_t StagefrightRecorder::setAudioSource(audio_source_t as) {
LOGV("setAudioSource: %d", as);
if (as < AUDIO_SOURCE_DEFAULT ||
- as >= AUDIO_SOURCE_LIST_END) {
+ as >= AUDIO_SOURCE_CNT) {
LOGE("Invalid audio source: %d", as);
return BAD_VALUE;
}
@@ -800,7 +802,7 @@
mStarted = true;
uint32_t params = IMediaPlayerService::kBatteryDataCodecStarted;
- if (mAudioSource != AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource != AUDIO_SOURCE_CNT) {
params |= IMediaPlayerService::kBatteryDataTrackAudio;
}
if (mVideoSource != VIDEO_SOURCE_LIST_END) {
@@ -874,7 +876,7 @@
mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
CHECK(mAudioEncoder == AUDIO_ENCODER_AAC);
- CHECK(mAudioSource != AUDIO_SOURCE_LIST_END);
+ CHECK(mAudioSource != AUDIO_SOURCE_CNT);
CHECK(0 == "AACWriter is not implemented yet");
@@ -900,7 +902,7 @@
}
}
- if (mAudioSource >= AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource >= AUDIO_SOURCE_CNT) {
LOGE("Invalid audio source: %d", mAudioSource);
return BAD_VALUE;
}
@@ -933,9 +935,9 @@
status_t StagefrightRecorder::startRTPRecording() {
CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_RTP_AVP);
- if ((mAudioSource != AUDIO_SOURCE_LIST_END
+ if ((mAudioSource != AUDIO_SOURCE_CNT
&& mVideoSource != VIDEO_SOURCE_LIST_END)
- || (mAudioSource == AUDIO_SOURCE_LIST_END
+ || (mAudioSource == AUDIO_SOURCE_CNT
&& mVideoSource == VIDEO_SOURCE_LIST_END)) {
// Must have exactly one source.
return BAD_VALUE;
@@ -947,7 +949,7 @@
sp<MediaSource> source;
- if (mAudioSource != AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource != AUDIO_SOURCE_CNT) {
source = createAudioSource();
} else {
@@ -975,7 +977,7 @@
sp<MediaWriter> writer = new MPEG2TSWriter(mOutputFd);
- if (mAudioSource != AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource != AUDIO_SOURCE_CNT) {
if (mAudioEncoder != AUDIO_ENCODER_AAC) {
return ERROR_UNSUPPORTED;
}
@@ -1383,7 +1385,7 @@
// Audio source is added at the end if it exists.
// This help make sure that the "recoding" sound is suppressed for
// camcorder applications in the recorded files.
- if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_LIST_END)) {
+ if (!mCaptureTimeLapse && (mAudioSource != AUDIO_SOURCE_CNT)) {
err = setupAudioEncoder(writer);
if (err != OK) return err;
*totalBitRate += mAudioBitRate;
@@ -1504,7 +1506,7 @@
mStarted = false;
uint32_t params = 0;
- if (mAudioSource != AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource != AUDIO_SOURCE_CNT) {
params |= IMediaPlayerService::kBatteryDataTrackAudio;
}
if (mVideoSource != VIDEO_SOURCE_LIST_END) {
@@ -1555,7 +1557,7 @@
mStarted = false;
uint32_t params = 0;
- if (mAudioSource != AUDIO_SOURCE_LIST_END) {
+ if (mAudioSource != AUDIO_SOURCE_CNT) {
params |= IMediaPlayerService::kBatteryDataTrackAudio;
}
if (mVideoSource != VIDEO_SOURCE_LIST_END) {
@@ -1581,7 +1583,7 @@
stop();
// No audio or video source by default
- mAudioSource = AUDIO_SOURCE_LIST_END;
+ mAudioSource = AUDIO_SOURCE_CNT;
mVideoSource = VIDEO_SOURCE_LIST_END;
// Default parameters
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 2c440c1..3d463ea 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -22,6 +22,8 @@
#include <camera/CameraParameters.h>
#include <utils/String8.h>
+#include <hardware/audio.h>
+
namespace android {
class Camera;
@@ -39,7 +41,7 @@
virtual ~StagefrightRecorder();
virtual status_t init();
- virtual status_t setAudioSource(audio_source as);
+ virtual status_t setAudioSource(audio_source_t as);
virtual status_t setVideoSource(video_source vs);
virtual status_t setOutputFormat(output_format of);
virtual status_t setAudioEncoder(audio_encoder ae);
@@ -69,7 +71,7 @@
sp<MediaWriter> mWriter, mWriterAux;
sp<AudioSource> mAudioSourceNode;
- audio_source mAudioSource;
+ audio_source_t mAudioSource;
video_source mVideoSource;
output_format mOutputFormat;
audio_encoder mAudioEncoder;
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index fcea848..69f9c23 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -110,7 +110,7 @@
if (mAudioSink.get() != NULL) {
status_t err = mAudioSink->open(
- mSampleRate, numChannels, AudioSystem::PCM_16_BIT,
+ mSampleRate, numChannels, AUDIO_FORMAT_PCM_16_BIT,
DEFAULT_AUDIOSINK_BUFFERCOUNT,
&AudioPlayer::AudioSinkCallback, this);
if (err != OK) {
@@ -132,10 +132,10 @@
mAudioSink->start();
} else {
mAudioTrack = new AudioTrack(
- AudioSystem::MUSIC, mSampleRate, AudioSystem::PCM_16_BIT,
+ AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT,
(numChannels == 2)
- ? AudioSystem::CHANNEL_OUT_STEREO
- : AudioSystem::CHANNEL_OUT_MONO,
+ ? AUDIO_CHANNEL_OUT_STEREO
+ : AUDIO_CHANNEL_OUT_MONO,
0, 0, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index bbdec02..99c3682 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -60,8 +60,8 @@
AudioRecord::RECORD_NS_ENABLE |
AudioRecord::RECORD_IIR_ENABLE;
mRecord = new AudioRecord(
- inputSource, sampleRate, AudioSystem::PCM_16_BIT,
- channels > 1? AudioSystem::CHANNEL_IN_STEREO: AudioSystem::CHANNEL_IN_MONO,
+ inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
+ channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO,
4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */
flags,
AudioRecordCallbackFunction,