Merge "Codec2: Initialize InputSurfaceWrapper::Config structure fields"
diff --git a/apex/Android.bp b/apex/Android.bp
index f3e8a55..6ba9cb9 100644
--- a/apex/Android.bp
+++ b/apex/Android.bp
@@ -97,6 +97,7 @@
"crash_dump.policy",
"mediaswcodec.xml",
],
+ use_vendor: true,
key: "com.android.media.swcodec.key",
certificate: ":com.android.media.swcodec.certificate",
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index f4b8164..e000633 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -272,7 +272,7 @@
status_t SimplePlayer::onPrepare() {
CHECK_EQ(mState, UNPREPARED);
- mExtractor = new NuMediaExtractor;
+ mExtractor = new NuMediaExtractor(NuMediaExtractor::EntryPoint::OTHER);
status_t err = mExtractor->setDataSource(
NULL /* httpService */, mPath.c_str());
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index c26e0b9..33c4663 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -79,7 +79,7 @@
static int64_t kTimeout = 500ll;
- sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor(NuMediaExtractor::EntryPoint::OTHER);
if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
fprintf(stderr, "unable to instantiate extractor.\n");
return 1;
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
index b894545..ca058ab 100644
--- a/cmds/stagefright/mediafilter.cpp
+++ b/cmds/stagefright/mediafilter.cpp
@@ -319,7 +319,8 @@
static int64_t kTimeout = 500ll;
- sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor(NuMediaExtractor::EntryPoint::OTHER);
+
if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
fprintf(stderr, "unable to instantiate extractor.\n");
return 1;
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index 4a83a4a..bc7e41e 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -62,7 +62,7 @@
int trimEndTimeMs,
int rotationDegrees,
MediaMuxer::OutputFormat container = MediaMuxer::OUTPUT_FORMAT_MPEG_4) {
- sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor(NuMediaExtractor::EntryPoint::OTHER);
if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
fprintf(stderr, "unable to instantiate extractor. %s\n", path);
return 1;
diff --git a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
index 12ed725..b520c17 100644
--- a/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/video/VtsHalMediaC2V1_0TargetVideoDecTest.cpp
@@ -734,7 +734,7 @@
}
if (timestampMax < timestamp) timestampMax = timestamp;
}
- timestampOffset = timestampMax;
+ timestampOffset = timestampMax + 33333;
eleInfo.close();
// Reset Total frames before second decode loop
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3ef454b..06464b5 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -1403,6 +1403,7 @@
continue;
}
if (work->input.buffers.empty()
+ || work->input.buffers.front() == nullptr
|| work->input.buffers.front()->data().linearBlocks().empty()) {
ALOGD("[%s] no linear codec config data found", mName);
continue;
diff --git a/media/codecs/g711/decoder/Android.bp b/media/codecs/g711/decoder/Android.bp
index 51f4c38..efff60b 100644
--- a/media/codecs/g711/decoder/Android.bp
+++ b/media/codecs/g711/decoder/Android.bp
@@ -35,13 +35,7 @@
],
cfi: true,
},
-
- apex_available: [
- "//apex_available:platform",
- "com.android.media.swcodec",
- "test_com.android.media.swcodec",
- ],
-
+ apex_available: ["com.android.media.swcodec"],
min_sdk_version: "29",
target: {
diff --git a/media/codecs/mp3dec/Android.bp b/media/codecs/mp3dec/Android.bp
index 1acf0a6..f84da21 100644
--- a/media/codecs/mp3dec/Android.bp
+++ b/media/codecs/mp3dec/Android.bp
@@ -1,3 +1,18 @@
+cc_library_headers {
+ name: "libstagefright_mp3dec_headers",
+ vendor_available: true,
+ min_sdk_version: "29",
+ host_supported:true,
+ export_include_dirs: [
+ "include",
+ "src",
+ ],
+ apex_available: [
+ "//apex_available:platform",
+ "com.android.media.swcodec",
+ ],
+}
+
cc_library_static {
name: "libstagefright_mp3dec",
vendor_available: true,
@@ -65,10 +80,8 @@
include_dirs: ["frameworks/av/media/libstagefright/include"],
- export_include_dirs: [
- "include",
- "src",
- ],
+ header_libs: ["libstagefright_mp3dec_headers"],
+ export_header_lib_headers: ["libstagefright_mp3dec_headers"],
cflags: [
"-DOSCL_UNUSED_ARG(x)=(void)(x)",
diff --git a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
index 2b2fe6d..6e3a1c8 100644
--- a/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderAdapter.cpp
@@ -15,10 +15,12 @@
*/
#include <binding/AAudioBinderAdapter.h>
+#include <media/AidlConversionUtil.h>
#include <utility/AAudioUtilities.h>
namespace aaudio {
+using android::aidl_utils::statusTFromBinderStatus;
using android::binder::Status;
AAudioBinderAdapter::AAudioBinderAdapter(IAAudioService* delegate)
@@ -36,7 +38,7 @@
¶ms,
&result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
config = params;
return result;
@@ -46,7 +48,7 @@
aaudio_result_t result;
Status status = mDelegate->closeStream(streamHandle, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -59,7 +61,7 @@
&endpoint,
&result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
endpointOut = std::move(endpoint);
return result;
@@ -69,7 +71,7 @@
aaudio_result_t result;
Status status = mDelegate->startStream(streamHandle, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -78,7 +80,7 @@
aaudio_result_t result;
Status status = mDelegate->pauseStream(streamHandle, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -87,7 +89,7 @@
aaudio_result_t result;
Status status = mDelegate->stopStream(streamHandle, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -96,7 +98,7 @@
aaudio_result_t result;
Status status = mDelegate->flushStream(streamHandle, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -107,7 +109,7 @@
aaudio_result_t result;
Status status = mDelegate->registerAudioThread(streamHandle, clientThreadId, periodNanoseconds, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
@@ -117,7 +119,7 @@
aaudio_result_t result;
Status status = mDelegate->unregisterAudioThread(streamHandle, clientThreadId, &result);
if (!status.isOk()) {
- result = AAudioConvert_androidToAAudioResult(status.transactionError());
+ result = AAudioConvert_androidToAAudioResult(statusTFromBinderStatus(status));
}
return result;
}
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1d036d0..af8ff19 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -558,7 +558,7 @@
if (status < 0) { // a non-negative value is the volume shaper id.
ALOGE("applyVolumeShaper() failed with status %d", status);
}
- return binder::Status::fromStatusT(status);
+ return aidl_utils::binderStatusFromStatusT(status);
} else {
ALOGD("applyVolumeShaper()"
" no AudioTrack for volume control from IPlayer");
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index a27cf78..7e5eba3 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -117,56 +117,58 @@
ConversionResult<Direction> direction(media::AudioPortRole role, media::AudioPortType type) {
switch (type) {
+ case media::AudioPortType::NONE:
+ case media::AudioPortType::SESSION:
+ break; // must be listed -Werror,-Wswitch
case media::AudioPortType::DEVICE:
switch (role) {
+ case media::AudioPortRole::NONE:
+ break; // must be listed -Werror,-Wswitch
case media::AudioPortRole::SOURCE:
return Direction::INPUT;
case media::AudioPortRole::SINK:
return Direction::OUTPUT;
- default:
- break;
}
break;
case media::AudioPortType::MIX:
switch (role) {
+ case media::AudioPortRole::NONE:
+ break; // must be listed -Werror,-Wswitch
case media::AudioPortRole::SOURCE:
return Direction::OUTPUT;
case media::AudioPortRole::SINK:
return Direction::INPUT;
- default:
- break;
}
break;
- default:
- break;
}
return unexpected(BAD_VALUE);
}
ConversionResult<Direction> direction(audio_port_role_t role, audio_port_type_t type) {
switch (type) {
+ case AUDIO_PORT_TYPE_NONE:
+ case AUDIO_PORT_TYPE_SESSION:
+ break; // must be listed -Werror,-Wswitch
case AUDIO_PORT_TYPE_DEVICE:
switch (role) {
+ case AUDIO_PORT_ROLE_NONE:
+ break; // must be listed -Werror,-Wswitch
case AUDIO_PORT_ROLE_SOURCE:
return Direction::INPUT;
case AUDIO_PORT_ROLE_SINK:
return Direction::OUTPUT;
- default:
- break;
}
break;
case AUDIO_PORT_TYPE_MIX:
switch (role) {
+ case AUDIO_PORT_ROLE_NONE:
+ break; // must be listed -Werror,-Wswitch
case AUDIO_PORT_ROLE_SOURCE:
return Direction::OUTPUT;
case AUDIO_PORT_ROLE_SINK:
return Direction::INPUT;
- default:
- break;
}
break;
- default:
- break;
}
return unexpected(BAD_VALUE);
}
@@ -276,8 +278,9 @@
return std::string(legacy.c_str());
}
-// The legacy enum is unnamed. Thus, we use int.
-ConversionResult<int> aidl2legacy_AudioPortConfigType(media::AudioPortConfigType aidl) {
+// The legacy enum is unnamed. Thus, we use int32_t.
+ConversionResult<int32_t> aidl2legacy_AudioPortConfigType_int32_t(
+ media::AudioPortConfigType aidl) {
switch (aidl) {
case media::AudioPortConfigType::SAMPLE_RATE:
return AUDIO_PORT_CONFIG_SAMPLE_RATE;
@@ -293,8 +296,9 @@
return unexpected(BAD_VALUE);
}
-// The legacy enum is unnamed. Thus, we use int.
-ConversionResult<media::AudioPortConfigType> legacy2aidl_AudioPortConfigType(int legacy) {
+// The legacy enum is unnamed. Thus, we use int32_t.
+ConversionResult<media::AudioPortConfigType> legacy2aidl_int32_t_AudioPortConfigType(
+ int32_t legacy) {
switch (legacy) {
case AUDIO_PORT_CONFIG_SAMPLE_RATE:
return media::AudioPortConfigType::SAMPLE_RATE;
@@ -312,7 +316,7 @@
ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl) {
return convertBitmask<unsigned int, int32_t, int, media::AudioPortConfigType>(
- aidl, aidl2legacy_AudioPortConfigType,
+ aidl, aidl2legacy_AudioPortConfigType_int32_t,
// AudioPortConfigType enum is index-based.
index2enum_index<media::AudioPortConfigType>,
// AUDIO_PORT_CONFIG_* flags are mask-based.
@@ -321,7 +325,7 @@
ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy) {
return convertBitmask<int32_t, unsigned int, media::AudioPortConfigType, int>(
- legacy, legacy2aidl_AudioPortConfigType,
+ legacy, legacy2aidl_int32_t_AudioPortConfigType,
// AUDIO_PORT_CONFIG_* flags are mask-based.
index2enum_bitmask<unsigned>,
// AudioPortConfigType enum is index-based.
@@ -363,9 +367,8 @@
return AUDIO_INPUT_CONFIG_CHANGED;
case media::AudioIoConfigEvent::CLIENT_STARTED:
return AUDIO_CLIENT_STARTED;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_AudioIoConfigEvent(
@@ -389,9 +392,8 @@
return media::AudioIoConfigEvent::INPUT_CONFIG_CHANGED;
case AUDIO_CLIENT_STARTED:
return media::AudioIoConfigEvent::CLIENT_STARTED;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_port_role_t> aidl2legacy_AudioPortRole_audio_port_role_t(
@@ -403,9 +405,8 @@
return AUDIO_PORT_ROLE_SOURCE;
case media::AudioPortRole::SINK:
return AUDIO_PORT_ROLE_SINK;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioPortRole> legacy2aidl_audio_port_role_t_AudioPortRole(
@@ -417,9 +418,8 @@
return media::AudioPortRole::SOURCE;
case AUDIO_PORT_ROLE_SINK:
return media::AudioPortRole::SINK;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_port_type_t> aidl2legacy_AudioPortType_audio_port_type_t(
@@ -433,9 +433,8 @@
return AUDIO_PORT_TYPE_MIX;
case media::AudioPortType::SESSION:
return AUDIO_PORT_TYPE_SESSION;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioPortType> legacy2aidl_audio_port_type_t_AudioPortType(
@@ -449,9 +448,8 @@
return media::AudioPortType::MIX;
case AUDIO_PORT_TYPE_SESSION:
return media::AudioPortType::SESSION;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_format_t> aidl2legacy_AudioFormat_audio_format_t(
@@ -476,9 +474,8 @@
return AUDIO_GAIN_MODE_CHANNELS;
case media::AudioGainMode::RAMP:
return AUDIO_GAIN_MODE_RAMP;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioGainMode> legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy) {
@@ -489,9 +486,8 @@
return media::AudioGainMode::CHANNELS;
case AUDIO_GAIN_MODE_RAMP:
return media::AudioGainMode::RAMP;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl) {
@@ -503,7 +499,7 @@
enumToMask_bitmask<audio_gain_mode_t, audio_gain_mode_t>);
}
-ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_mask_int32_t(audio_gain_mode_t legacy) {
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t_mask(audio_gain_mode_t legacy) {
return convertBitmask<int32_t, audio_gain_mode_t, media::AudioGainMode, audio_gain_mode_t>(
legacy, legacy2aidl_audio_gain_mode_t_AudioGainMode,
// AUDIO_GAIN_MODE_* constants are mask-based.
@@ -548,7 +544,7 @@
const audio_gain_config& legacy, audio_port_role_t role, audio_port_type_t type) {
media::AudioGainConfig aidl;
aidl.index = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.index));
- aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_mask_int32_t(legacy.mode));
+ aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
aidl.channelMask =
VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
@@ -583,14 +579,15 @@
return AUDIO_INPUT_FLAG_HW_AV_SYNC;
case media::AudioInputFlags::DIRECT:
return AUDIO_INPUT_FLAG_DIRECT;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioInputFlags> legacy2aidl_audio_input_flags_t_AudioInputFlags(
audio_input_flags_t legacy) {
switch (legacy) {
+ case AUDIO_INPUT_FLAG_NONE:
+ break; // shouldn't get here. must be listed -Werror,-Wswitch
case AUDIO_INPUT_FLAG_FAST:
return media::AudioInputFlags::FAST;
case AUDIO_INPUT_FLAG_HW_HOTWORD:
@@ -607,9 +604,8 @@
return media::AudioInputFlags::HW_AV_SYNC;
case AUDIO_INPUT_FLAG_DIRECT:
return media::AudioInputFlags::DIRECT;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_output_flags_t> aidl2legacy_AudioOutputFlags_audio_output_flags_t(
@@ -647,14 +643,15 @@
return AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
case media::AudioOutputFlags::GAPLESS_OFFLOAD:
return AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioOutputFlags> legacy2aidl_audio_output_flags_t_AudioOutputFlags(
audio_output_flags_t legacy) {
switch (legacy) {
+ case AUDIO_OUTPUT_FLAG_NONE:
+ break; // shouldn't get here. must be listed -Werror,-Wswitch
case AUDIO_OUTPUT_FLAG_DIRECT:
return media::AudioOutputFlags::DIRECT;
case AUDIO_OUTPUT_FLAG_PRIMARY:
@@ -687,12 +684,12 @@
return media::AudioOutputFlags::INCALL_MUSIC;
case AUDIO_OUTPUT_FLAG_GAPLESS_OFFLOAD:
return media::AudioOutputFlags::GAPLESS_OFFLOAD;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
-ConversionResult<audio_input_flags_t> aidl2legacy_audio_input_flags_mask(int32_t aidl) {
+ConversionResult<audio_input_flags_t> aidl2legacy_int32_t_audio_input_flags_t_mask(
+ int32_t aidl) {
using LegacyMask = std::underlying_type_t<audio_input_flags_t>;
LegacyMask converted = VALUE_OR_RETURN(
@@ -703,7 +700,8 @@
return static_cast<audio_input_flags_t>(converted);
}
-ConversionResult<int32_t> legacy2aidl_audio_input_flags_mask(audio_input_flags_t legacy) {
+ConversionResult<int32_t> legacy2aidl_audio_input_flags_t_int32_t_mask(
+ audio_input_flags_t legacy) {
using LegacyMask = std::underlying_type_t<audio_input_flags_t>;
LegacyMask legacyMask = static_cast<LegacyMask>(legacy);
@@ -713,7 +711,8 @@
enumToMask_index<int32_t, media::AudioInputFlags>);
}
-ConversionResult<audio_output_flags_t> aidl2legacy_audio_output_flags_mask(int32_t aidl) {
+ConversionResult<audio_output_flags_t> aidl2legacy_int32_t_audio_output_flags_t_mask(
+ int32_t aidl) {
return convertBitmask<audio_output_flags_t,
int32_t,
audio_output_flags_t,
@@ -723,7 +722,8 @@
enumToMask_bitmask<audio_output_flags_t, audio_output_flags_t>);
}
-ConversionResult<int32_t> legacy2aidl_audio_output_flags_mask(audio_output_flags_t legacy) {
+ConversionResult<int32_t> legacy2aidl_audio_output_flags_t_int32_t_mask(
+ audio_output_flags_t legacy) {
using LegacyMask = std::underlying_type_t<audio_output_flags_t>;
LegacyMask legacyMask = static_cast<LegacyMask>(legacy);
@@ -740,13 +740,15 @@
switch (dir) {
case Direction::INPUT: {
legacy.input = VALUE_OR_RETURN(
- aidl2legacy_audio_input_flags_mask(VALUE_OR_RETURN(UNION_GET(aidl, input))));
+ aidl2legacy_int32_t_audio_input_flags_t_mask(
+ VALUE_OR_RETURN(UNION_GET(aidl, input))));
}
break;
case Direction::OUTPUT: {
legacy.output = VALUE_OR_RETURN(
- aidl2legacy_audio_output_flags_mask(VALUE_OR_RETURN(UNION_GET(aidl, output))));
+ aidl2legacy_int32_t_audio_output_flags_t_mask(
+ VALUE_OR_RETURN(UNION_GET(aidl, output))));
}
break;
}
@@ -762,17 +764,20 @@
switch (dir) {
case Direction::INPUT:
UNION_SET(aidl, input,
- VALUE_OR_RETURN(legacy2aidl_audio_input_flags_mask(legacy.input)));
+ VALUE_OR_RETURN(legacy2aidl_audio_input_flags_t_int32_t_mask(
+ legacy.input)));
break;
case Direction::OUTPUT:
UNION_SET(aidl, output,
- VALUE_OR_RETURN(legacy2aidl_audio_output_flags_mask(legacy.output)));
+ VALUE_OR_RETURN(legacy2aidl_audio_output_flags_t_int32_t_mask(
+ legacy.output)));
break;
}
return aidl;
}
-ConversionResult<audio_port_config_device_ext> aidl2legacy_AudioPortConfigDeviceExt(
+ConversionResult<audio_port_config_device_ext>
+aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
const media::AudioPortConfigDeviceExt& aidl) {
audio_port_config_device_ext legacy;
legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
@@ -781,7 +786,8 @@
return legacy;
}
-ConversionResult<media::AudioPortConfigDeviceExt> legacy2aidl_AudioPortConfigDeviceExt(
+ConversionResult<media::AudioPortConfigDeviceExt>
+legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
const audio_port_config_device_ext& legacy) {
media::AudioPortConfigDeviceExt aidl;
aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
@@ -826,9 +832,8 @@
return AUDIO_STREAM_PATCH;
case media::AudioStreamType::CALL_ASSISTANT:
return AUDIO_STREAM_CALL_ASSISTANT;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioStreamType> legacy2aidl_audio_stream_type_t_AudioStreamType(
@@ -866,9 +871,8 @@
return media::AudioStreamType::PATCH;
case AUDIO_STREAM_CALL_ASSISTANT:
return media::AudioStreamType::CALL_ASSISTANT;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_source_t> aidl2legacy_AudioSourceType_audio_source_t(
@@ -905,9 +909,8 @@
return AUDIO_SOURCE_FM_TUNER;
case media::AudioSourceType::HOTWORD:
return AUDIO_SOURCE_HOTWORD;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<media::AudioSourceType> legacy2aidl_audio_source_t_AudioSourceType(
@@ -943,9 +946,8 @@
return media::AudioSourceType::FM_TUNER;
case AUDIO_SOURCE_HOTWORD:
return media::AudioSourceType::HOTWORD;
- default:
- return unexpected(BAD_VALUE);
}
+ return unexpected(BAD_VALUE);
}
ConversionResult<audio_session_t> aidl2legacy_int32_t_audio_session_t(int32_t aidl) {
@@ -967,24 +969,21 @@
case media::AudioPortRole::NONE:
// Just verify that the union is empty.
VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
- break;
+ return legacy;
case media::AudioPortRole::SOURCE:
// This is not a bug. A SOURCE role corresponds to the stream field.
legacy.stream = VALUE_OR_RETURN(aidl2legacy_AudioStreamType_audio_stream_type_t(
VALUE_OR_RETURN(UNION_GET(aidl, stream))));
- break;
+ return legacy;
case media::AudioPortRole::SINK:
// This is not a bug. A SINK role corresponds to the source field.
legacy.source = VALUE_OR_RETURN(aidl2legacy_AudioSourceType_audio_source_t(
VALUE_OR_RETURN(UNION_GET(aidl, source))));
- break;
-
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ return legacy;
}
- return legacy;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<media::AudioPortConfigMixExtUseCase> legacy2aidl_AudioPortConfigMixExtUseCase(
@@ -994,21 +993,19 @@
switch (role) {
case AUDIO_PORT_ROLE_NONE:
UNION_SET(aidl, unspecified, false);
- break;
+ return aidl;
case AUDIO_PORT_ROLE_SOURCE:
// This is not a bug. A SOURCE role corresponds to the stream field.
UNION_SET(aidl, stream, VALUE_OR_RETURN(
legacy2aidl_audio_stream_type_t_AudioStreamType(legacy.stream)));
- break;
+ return aidl;
case AUDIO_PORT_ROLE_SINK:
// This is not a bug. A SINK role corresponds to the source field.
UNION_SET(aidl, source,
VALUE_OR_RETURN(legacy2aidl_audio_source_t_AudioSourceType(legacy.source)));
- break;
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ return aidl;
}
- return aidl;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<audio_port_config_mix_ext> aidl2legacy_AudioPortConfigMixExt(
@@ -1029,14 +1026,16 @@
return aidl;
}
-ConversionResult<audio_port_config_session_ext> aidl2legacy_AudioPortConfigSessionExt(
+ConversionResult<audio_port_config_session_ext>
+aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
const media::AudioPortConfigSessionExt& aidl) {
audio_port_config_session_ext legacy;
legacy.session = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_session_t(aidl.session));
return legacy;
}
-ConversionResult<media::AudioPortConfigSessionExt> legacy2aidl_AudioPortConfigSessionExt(
+ConversionResult<media::AudioPortConfigSessionExt>
+legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
const audio_port_config_session_ext& legacy) {
media::AudioPortConfigSessionExt aidl;
aidl.session = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(legacy.session));
@@ -1054,23 +1053,24 @@
case media::AudioPortType::NONE:
// Just verify that the union is empty.
VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
- break;
+ return legacy;
case media::AudioPortType::DEVICE:
legacy.device = VALUE_OR_RETURN(
- aidl2legacy_AudioPortConfigDeviceExt(VALUE_OR_RETURN(UNION_GET(aidl, device))));
- break;
+ aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
+ VALUE_OR_RETURN(UNION_GET(aidl, device))));
+ return legacy;
case media::AudioPortType::MIX:
legacy.mix = VALUE_OR_RETURN(
aidl2legacy_AudioPortConfigMixExt(VALUE_OR_RETURN(UNION_GET(aidl, mix)), role));
- break;
+ return legacy;
case media::AudioPortType::SESSION:
- legacy.session = VALUE_OR_RETURN(aidl2legacy_AudioPortConfigSessionExt(
- VALUE_OR_RETURN(UNION_GET(aidl, session))));
- break;
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ legacy.session = VALUE_OR_RETURN(
+ aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
+ VALUE_OR_RETURN(UNION_GET(aidl, session))));
+ return legacy;
+
}
- return legacy;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<media::AudioPortConfigExt> legacy2aidl_AudioPortConfigExt(
@@ -1080,23 +1080,25 @@
switch (type) {
case AUDIO_PORT_TYPE_NONE:
UNION_SET(aidl, unspecified, false);
- break;
+ return aidl;
case AUDIO_PORT_TYPE_DEVICE:
UNION_SET(aidl, device,
- VALUE_OR_RETURN(legacy2aidl_AudioPortConfigDeviceExt(legacy.device)));
- break;
+ VALUE_OR_RETURN(
+ legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
+ legacy.device)));
+ return aidl;
case AUDIO_PORT_TYPE_MIX:
UNION_SET(aidl, mix,
VALUE_OR_RETURN(legacy2aidl_AudioPortConfigMixExt(legacy.mix, role)));
- break;
+ return aidl;
case AUDIO_PORT_TYPE_SESSION:
UNION_SET(aidl, session,
- VALUE_OR_RETURN(legacy2aidl_AudioPortConfigSessionExt(legacy.session)));
- break;
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ VALUE_OR_RETURN(
+ legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
+ legacy.session)));
+ return aidl;
}
- return aidl;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
@@ -1235,7 +1237,8 @@
return aidl;
}
-ConversionResult<AudioClient> aidl2legacy_AudioClient(const media::AudioClient& aidl) {
+ConversionResult<AudioClient> aidl2legacy_AudioClient_AudioClient(
+ const media::AudioClient& aidl) {
AudioClient legacy;
legacy.clientUid = VALUE_OR_RETURN(aidl2legacy_int32_t_uid_t(aidl.clientUid));
legacy.clientPid = VALUE_OR_RETURN(aidl2legacy_int32_t_pid_t(aidl.clientPid));
@@ -1244,7 +1247,8 @@
return legacy;
}
-ConversionResult<media::AudioClient> legacy2aidl_AudioClient(const AudioClient& legacy) {
+ConversionResult<media::AudioClient> legacy2aidl_AudioClient_AudioClient(
+ const AudioClient& legacy) {
media::AudioClient aidl;
aidl.clientUid = VALUE_OR_RETURN(legacy2aidl_uid_t_int32_t(legacy.clientUid));
aidl.clientPid = VALUE_OR_RETURN(legacy2aidl_pid_t_int32_t(legacy.clientPid));
@@ -1672,7 +1676,7 @@
}
ConversionResult<AudioTimestamp>
-aidl2legacy_AudioTimestamp(const media::AudioTimestampInternal& aidl) {
+aidl2legacy_AudioTimestampInternal_AudioTimestamp(const media::AudioTimestampInternal& aidl) {
AudioTimestamp legacy;
legacy.mPosition = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.position));
legacy.mTime.tv_sec = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sec));
@@ -1681,7 +1685,7 @@
}
ConversionResult<media::AudioTimestampInternal>
-legacy2aidl_AudioTimestamp(const AudioTimestamp& legacy) {
+legacy2aidl_AudioTimestamp_AudioTimestampInternal(const AudioTimestamp& legacy) {
media::AudioTimestampInternal aidl;
aidl.position = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.mPosition));
aidl.sec = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy.mTime.tv_sec));
@@ -1912,25 +1916,24 @@
case media::AudioPortType::NONE:
// Just verify that the union is empty.
VALUE_OR_RETURN(UNION_GET(aidl, unspecified));
- break;
+ return legacy;
case media::AudioPortType::DEVICE:
legacy.device = VALUE_OR_RETURN(
aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(
VALUE_OR_RETURN(UNION_GET(aidl, device))));
- break;
+ return legacy;
case media::AudioPortType::MIX:
legacy.mix = VALUE_OR_RETURN(
aidl2legacy_AudioPortMixExt_audio_port_mix_ext(
VALUE_OR_RETURN(UNION_GET(aidl, mix))));
- break;
+ return legacy;
case media::AudioPortType::SESSION:
legacy.session = VALUE_OR_RETURN(aidl2legacy_AudioPortSessionExt_audio_port_session_ext(
VALUE_OR_RETURN(UNION_GET(aidl, session))));
- break;
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ return legacy;
+
}
- return legacy;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<media::AudioPortExt> legacy2aidl_AudioPortExt(
@@ -1939,25 +1942,23 @@
switch (type) {
case AUDIO_PORT_TYPE_NONE:
UNION_SET(aidl, unspecified, false);
- break;
+ return aidl;
case AUDIO_PORT_TYPE_DEVICE:
UNION_SET(aidl, device,
VALUE_OR_RETURN(
legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(legacy.device)));
- break;
+ return aidl;
case AUDIO_PORT_TYPE_MIX:
UNION_SET(aidl, mix,
VALUE_OR_RETURN(legacy2aidl_audio_port_mix_ext_AudioPortMixExt(legacy.mix)));
- break;
+ return aidl;
case AUDIO_PORT_TYPE_SESSION:
UNION_SET(aidl, session,
VALUE_OR_RETURN(legacy2aidl_audio_port_session_ext_AudioPortSessionExt(
legacy.session)));
- break;
- default:
- LOG_ALWAYS_FATAL("Shouldn't get here");
+ return aidl;
}
- return aidl;
+ LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
}
ConversionResult<audio_profile>
@@ -2024,7 +2025,7 @@
ConversionResult<media::AudioGain>
legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy) {
media::AudioGain aidl;
- aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_mask_int32_t(legacy.mode));
+ aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
aidl.channelMask = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
aidl.minValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.min_value));
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index ae899c0..79ea1bb 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -30,7 +30,7 @@
#include <utils/Log.h>
namespace android {
-
+using aidl_utils::statusTFromBinderStatus;
using binder::Status;
namespace {
@@ -262,7 +262,7 @@
bs = mIEffect->disable(&status);
}
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
}
if (status == NO_ERROR) {
mEnabled = enabled;
@@ -303,7 +303,7 @@
Status bs = mIEffect->command(cmdCode, data, *replySize, &response, &status);
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
}
if (status == NO_ERROR) {
memcpy(replyData, response.data(), response.size());
@@ -351,7 +351,7 @@
&response,
&status);
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
return status;
}
assert(response.size() == sizeof(int));
@@ -410,7 +410,7 @@
&response,
&status);
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
}
return status;
}
@@ -441,7 +441,7 @@
Status bs = mIEffect->command(EFFECT_CMD_GET_PARAM, cmd, psize, &response, &status);
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
return status;
}
memcpy(param, response.data(), response.size());
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index f01b1d0..112cb67 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -47,6 +47,8 @@
#define WAIT_PERIOD_MS 10
namespace android {
+using aidl_utils::statusTFromBinderStatus;
+
// ---------------------------------------------------------------------------
// static
@@ -450,7 +452,7 @@
mActive = true;
if (!(flags & CBLK_INVALID)) {
- status = mAudioRecord->start(event, triggerSession).transactionError();
+ status = statusTFromBinderStatus(mAudioRecord->start(event, triggerSession));
if (status == DEAD_OBJECT) {
flags |= CBLK_INVALID;
}
@@ -1439,8 +1441,8 @@
if (mActive) {
// callback thread or sync event hasn't changed
// FIXME this fails if we have a new AudioFlinger instance
- result = mAudioRecord->start(
- AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE).transactionError();
+ result = statusTFromBinderStatus(mAudioRecord->start(
+ AudioSystem::SYNC_EVENT_SAME, AUDIO_SESSION_NONE));
}
mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset.
}
@@ -1531,7 +1533,7 @@
{
AutoMutex lock(mLock);
std::vector<media::MicrophoneInfoData> mics;
- status_t status = mAudioRecord->getActiveMicrophones(&mics).transactionError();
+ status_t status = statusTFromBinderStatus(mAudioRecord->getActiveMicrophones(&mics));
activeMicrophones->resize(mics.size());
for (size_t i = 0; status == OK && i < mics.size(); ++i) {
status = activeMicrophones->at(i).readFromParcelable(mics[i]);
@@ -1552,7 +1554,7 @@
// the internal AudioRecord hasn't be created yet, so just stash the attribute.
return OK;
} else {
- return mAudioRecord->setPreferredMicrophoneDirection(direction).transactionError();
+ return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneDirection(direction));
}
}
@@ -1568,7 +1570,7 @@
// the internal AudioRecord hasn't be created yet, so just stash the attribute.
return OK;
} else {
- return mAudioRecord->setPreferredMicrophoneFieldDimension(zoom).transactionError();
+ return statusTFromBinderStatus(mAudioRecord->setPreferredMicrophoneFieldDimension(zoom));
}
}
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 9aef794..09fcc66 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -35,7 +35,7 @@
#define VALUE_OR_RETURN_BINDER_STATUS(x) \
({ auto _tmp = (x); \
- if (!_tmp.ok()) return Status::fromStatusT(_tmp.error()); \
+ if (!_tmp.ok()) return aidl_utils::binderStatusFromStatusT(_tmp.error()); \
std::move(_tmp.value()); })
// ----------------------------------------------------------------------------
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 8c53c5b..1b1e143 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -2819,7 +2819,7 @@
media::AudioTimestampInternal ts;
mAudioTrack->getTimestamp(&ts, &status);
if (status == OK) {
- timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestamp(ts));
+ timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
}
} else {
// read timestamp from shared memory
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index a7cca45..20124df 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -28,6 +28,7 @@
namespace android {
+using aidl_utils::statusTFromBinderStatus;
using binder::Status;
#define MAX_ITEMS_PER_LIST 1024
@@ -55,13 +56,13 @@
media::CreateTrackRequest aidl;
aidl.attr = VALUE_OR_RETURN(legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
aidl.config = VALUE_OR_RETURN(legacy2aidl_audio_config_t_AudioConfig(config));
- aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient(clientInfo));
+ aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient_AudioClient(clientInfo));
aidl.sharedBuffer = VALUE_OR_RETURN(legacy2aidl_NullableIMemory_SharedFileRegion(sharedBuffer));
aidl.notificationsPerBuffer = VALUE_OR_RETURN(convertIntegral<int32_t>(notificationsPerBuffer));
aidl.speed = speed;
aidl.audioTrackCallback = audioTrackCallback;
aidl.opPackageName = opPackageName;
- aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_output_flags_mask(flags));
+ aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(frameCount));
aidl.notificationFrameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(notificationFrameCount));
aidl.selectedDeviceId = VALUE_OR_RETURN(
@@ -75,14 +76,14 @@
IAudioFlinger::CreateTrackInput legacy;
legacy.attr = VALUE_OR_RETURN(aidl2legacy_AudioAttributesInternal_audio_attributes_t(aidl.attr));
legacy.config = VALUE_OR_RETURN(aidl2legacy_AudioConfig_audio_config_t(aidl.config));
- legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient(aidl.clientInfo));
+ legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient_AudioClient(aidl.clientInfo));
legacy.sharedBuffer = VALUE_OR_RETURN(aidl2legacy_NullableSharedFileRegion_IMemory(aidl.sharedBuffer));
legacy.notificationsPerBuffer = VALUE_OR_RETURN(
convertIntegral<uint32_t>(aidl.notificationsPerBuffer));
legacy.speed = aidl.speed;
legacy.audioTrackCallback = aidl.audioTrackCallback;
legacy.opPackageName = aidl.opPackageName;
- legacy.flags = VALUE_OR_RETURN(aidl2legacy_audio_output_flags_mask(aidl.flags));
+ legacy.flags = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_output_flags_t_mask(aidl.flags));
legacy.frameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
legacy.notificationFrameCount = VALUE_OR_RETURN(
convertIntegral<size_t>(aidl.notificationFrameCount));
@@ -95,7 +96,7 @@
ConversionResult<media::CreateTrackResponse>
IAudioFlinger::CreateTrackOutput::toAidl() const {
media::CreateTrackResponse aidl;
- aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_output_flags_mask(flags));
+ aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(frameCount));
aidl.notificationFrameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(notificationFrameCount));
aidl.selectedDeviceId = VALUE_OR_RETURN(
@@ -115,7 +116,7 @@
IAudioFlinger::CreateTrackOutput::fromAidl(
const media::CreateTrackResponse& aidl) {
IAudioFlinger::CreateTrackOutput legacy;
- legacy.flags = VALUE_OR_RETURN(aidl2legacy_audio_output_flags_mask(aidl.flags));
+ legacy.flags = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_output_flags_t_mask(aidl.flags));
legacy.frameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
legacy.notificationFrameCount = VALUE_OR_RETURN(
convertIntegral<size_t>(aidl.notificationFrameCount));
@@ -137,10 +138,10 @@
media::CreateRecordRequest aidl;
aidl.attr = VALUE_OR_RETURN(legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
aidl.config = VALUE_OR_RETURN(legacy2aidl_audio_config_base_t_AudioConfigBase(config));
- aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient(clientInfo));
+ aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient_AudioClient(clientInfo));
aidl.opPackageName = VALUE_OR_RETURN(legacy2aidl_String16_string(opPackageName));
aidl.riid = VALUE_OR_RETURN(legacy2aidl_audio_unique_id_t_int32_t(riid));
- aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_input_flags_mask(flags));
+ aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(frameCount));
aidl.notificationFrameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(notificationFrameCount));
aidl.selectedDeviceId = VALUE_OR_RETURN(
@@ -155,10 +156,10 @@
IAudioFlinger::CreateRecordInput legacy;
legacy.attr = VALUE_OR_RETURN(aidl2legacy_AudioAttributesInternal_audio_attributes_t(aidl.attr));
legacy.config = VALUE_OR_RETURN(aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config));
- legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient(aidl.clientInfo));
+ legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient_AudioClient(aidl.clientInfo));
legacy.opPackageName = VALUE_OR_RETURN(aidl2legacy_string_view_String16(aidl.opPackageName));
legacy.riid = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_unique_id_t(aidl.riid));
- legacy.flags = VALUE_OR_RETURN(aidl2legacy_audio_input_flags_mask(aidl.flags));
+ legacy.flags = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_input_flags_t_mask(aidl.flags));
legacy.frameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
legacy.notificationFrameCount = VALUE_OR_RETURN(
convertIntegral<size_t>(aidl.notificationFrameCount));
@@ -171,7 +172,7 @@
ConversionResult<media::CreateRecordResponse>
IAudioFlinger::CreateRecordOutput::toAidl() const {
media::CreateRecordResponse aidl;
- aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_input_flags_mask(flags));
+ aidl.flags = VALUE_OR_RETURN(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(frameCount));
aidl.notificationFrameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(notificationFrameCount));
aidl.selectedDeviceId = VALUE_OR_RETURN(
@@ -190,7 +191,7 @@
IAudioFlinger::CreateRecordOutput::fromAidl(
const media::CreateRecordResponse& aidl) {
IAudioFlinger::CreateRecordOutput legacy;
- legacy.flags = VALUE_OR_RETURN(aidl2legacy_audio_input_flags_mask(aidl.flags));
+ legacy.flags = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_input_flags_t_mask(aidl.flags));
legacy.frameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
legacy.notificationFrameCount = VALUE_OR_RETURN(
convertIntegral<size_t>(aidl.notificationFrameCount));
@@ -214,19 +215,19 @@
status_t AudioFlingerClientAdapter::createTrack(const media::CreateTrackRequest& input,
media::CreateTrackResponse& output) {
- return mDelegate->createTrack(input, &output).transactionError();
+ return statusTFromBinderStatus(mDelegate->createTrack(input, &output));
}
status_t AudioFlingerClientAdapter::createRecord(const media::CreateRecordRequest& input,
media::CreateRecordResponse& output) {
- return mDelegate->createRecord(input, &output).transactionError();
+ return statusTFromBinderStatus(mDelegate->createRecord(input, &output));
}
uint32_t AudioFlingerClientAdapter::sampleRate(audio_io_handle_t ioHandle) const {
auto result = [&]() -> ConversionResult<uint32_t> {
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->sampleRate(ioHandleAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->sampleRate(ioHandleAidl, &aidlRet)));
return convertIntegral<uint32_t>(aidlRet);
}();
// Failure is ignored.
@@ -237,7 +238,7 @@
auto result = [&]() -> ConversionResult<audio_format_t> {
int32_t outputAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output));
media::audio::common::AudioFormat aidlRet;
- RETURN_IF_ERROR(mDelegate->format(outputAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->format(outputAidl, &aidlRet)));
return aidl2legacy_AudioFormat_audio_format_t(aidlRet);
}();
return result.value_or(AUDIO_FORMAT_INVALID);
@@ -247,7 +248,7 @@
auto result = [&]() -> ConversionResult<size_t> {
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
int64_t aidlRet;
- RETURN_IF_ERROR(mDelegate->frameCount(ioHandleAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->frameCount(ioHandleAidl, &aidlRet)));
return convertIntegral<size_t>(aidlRet);
}();
// Failure is ignored.
@@ -258,7 +259,7 @@
auto result = [&]() -> ConversionResult<uint32_t> {
int32_t outputAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output));
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->latency(outputAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->latency(outputAidl, &aidlRet)));
return convertIntegral<uint32_t>(aidlRet);
}();
// Failure is ignored.
@@ -266,17 +267,17 @@
}
status_t AudioFlingerClientAdapter::setMasterVolume(float value) {
- return mDelegate->setMasterVolume(value).transactionError();
+ return statusTFromBinderStatus(mDelegate->setMasterVolume(value));
}
status_t AudioFlingerClientAdapter::setMasterMute(bool muted) {
- return mDelegate->setMasterMute(muted).transactionError();
+ return statusTFromBinderStatus(mDelegate->setMasterMute(muted));
}
float AudioFlingerClientAdapter::masterVolume() const {
auto result = [&]() -> ConversionResult<float> {
float aidlRet;
- RETURN_IF_ERROR(mDelegate->masterVolume(&aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->masterVolume(&aidlRet)));
return aidlRet;
}();
// Failure is ignored.
@@ -286,7 +287,7 @@
bool AudioFlingerClientAdapter::masterMute() const {
auto result = [&]() -> ConversionResult<bool> {
bool aidlRet;
- RETURN_IF_ERROR(mDelegate->masterMute(&aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->masterMute(&aidlRet)));
return aidlRet;
}();
// Failure is ignored.
@@ -294,11 +295,11 @@
}
status_t AudioFlingerClientAdapter::setMasterBalance(float balance) {
- return mDelegate->setMasterBalance(balance).transactionError();
+ return statusTFromBinderStatus(mDelegate->setMasterBalance(balance));
}
status_t AudioFlingerClientAdapter::getMasterBalance(float* balance) const{
- return mDelegate->getMasterBalance(balance).transactionError();
+ return statusTFromBinderStatus(mDelegate->getMasterBalance(balance));
}
status_t AudioFlingerClientAdapter::setStreamVolume(audio_stream_type_t stream, float value,
@@ -306,13 +307,13 @@
media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- return mDelegate->setStreamVolume(streamAidl, value, outputAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->setStreamVolume(streamAidl, value, outputAidl));
}
status_t AudioFlingerClientAdapter::setStreamMute(audio_stream_type_t stream, bool muted) {
media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
- return mDelegate->setStreamMute(streamAidl, muted).transactionError();
+ return statusTFromBinderStatus(mDelegate->setStreamMute(streamAidl, muted));
}
float AudioFlingerClientAdapter::streamVolume(audio_stream_type_t stream,
@@ -322,8 +323,8 @@
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
float aidlRet;
- RETURN_IF_ERROR(
- mDelegate->streamVolume(streamAidl, outputAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->streamVolume(streamAidl, outputAidl, &aidlRet)));
return aidlRet;
}();
// Failure is ignored.
@@ -335,8 +336,8 @@
media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
bool aidlRet;
- RETURN_IF_ERROR(
- mDelegate->streamMute(streamAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->streamMute(streamAidl, &aidlRet)));
return aidlRet;
}();
// Failure is ignored.
@@ -345,18 +346,18 @@
status_t AudioFlingerClientAdapter::setMode(audio_mode_t mode) {
media::AudioMode modeAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_mode_t_AudioMode(mode));
- return mDelegate->setMode(modeAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->setMode(modeAidl));
}
status_t AudioFlingerClientAdapter::setMicMute(bool state) {
- return mDelegate->setMicMute(state).transactionError();
+ return statusTFromBinderStatus(mDelegate->setMicMute(state));
}
bool AudioFlingerClientAdapter::getMicMute() const {
auto result = [&]() -> ConversionResult<bool> {
bool aidlRet;
- RETURN_IF_ERROR(
- mDelegate->getMicMute(&aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getMicMute(&aidlRet)));
return aidlRet;
}();
// Failure is ignored.
@@ -367,7 +368,7 @@
auto result = [&]() -> status_t {
int32_t portIdAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_port_handle_t_int32_t(portId));
- return mDelegate->setRecordSilenced(portIdAidl, silenced).transactionError();
+ return statusTFromBinderStatus(mDelegate->setRecordSilenced(portIdAidl, silenced));
}();
// Failure is ignored.
(void) result;
@@ -378,7 +379,7 @@
int32_t ioHandleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
std::string keyValuePairsAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_String8_string(keyValuePairs));
- return mDelegate->setParameters(ioHandleAidl, keyValuePairsAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->setParameters(ioHandleAidl, keyValuePairsAidl));
}
String8 AudioFlingerClientAdapter::getParameters(audio_io_handle_t ioHandle, const String8& keys)
@@ -387,8 +388,8 @@
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
std::string keysAidl = VALUE_OR_RETURN(legacy2aidl_String8_string(keys));
std::string aidlRet;
- RETURN_IF_ERROR(
- mDelegate->getParameters(ioHandleAidl, keysAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getParameters(ioHandleAidl, keysAidl, &aidlRet)));
return aidl2legacy_string_view_String8(aidlRet);
}();
// Failure is ignored.
@@ -409,9 +410,9 @@
int32_t channelMaskAidl = VALUE_OR_RETURN(
legacy2aidl_audio_channel_mask_t_int32_t(channelMask));
int64_t aidlRet;
- RETURN_IF_ERROR(
+ RETURN_IF_ERROR(statusTFromBinderStatus(
mDelegate->getInputBufferSize(sampleRateAidl, formatAidl, channelMaskAidl,
- &aidlRet).transactionError());
+ &aidlRet)));
return convertIntegral<size_t>(aidlRet);
}();
// Failure is ignored.
@@ -420,7 +421,7 @@
status_t AudioFlingerClientAdapter::openOutput(const media::OpenOutputRequest& request,
media::OpenOutputResponse* response) {
- return mDelegate->openOutput(request, response).transactionError();
+ return statusTFromBinderStatus(mDelegate->openOutput(request, response));
}
audio_io_handle_t AudioFlingerClientAdapter::openDuplicateOutput(audio_io_handle_t output1,
@@ -429,8 +430,8 @@
int32_t output1Aidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output1));
int32_t output2Aidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output2));
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->openDuplicateOutput(output1Aidl, output2Aidl,
- &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->openDuplicateOutput(output1Aidl, output2Aidl, &aidlRet)));
return aidl2legacy_int32_t_audio_io_handle_t(aidlRet);
}();
// Failure is ignored.
@@ -439,44 +440,45 @@
status_t AudioFlingerClientAdapter::closeOutput(audio_io_handle_t output) {
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- return mDelegate->closeOutput(outputAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->closeOutput(outputAidl));
}
status_t AudioFlingerClientAdapter::suspendOutput(audio_io_handle_t output) {
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- return mDelegate->suspendOutput(outputAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->suspendOutput(outputAidl));
}
status_t AudioFlingerClientAdapter::restoreOutput(audio_io_handle_t output) {
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- return mDelegate->restoreOutput(outputAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->restoreOutput(outputAidl));
}
status_t AudioFlingerClientAdapter::openInput(const media::OpenInputRequest& request,
media::OpenInputResponse* response) {
- return mDelegate->openInput(request, response).transactionError();
+ return statusTFromBinderStatus(mDelegate->openInput(request, response));
}
status_t AudioFlingerClientAdapter::closeInput(audio_io_handle_t input) {
int32_t inputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
- return mDelegate->closeInput(inputAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->closeInput(inputAidl));
}
status_t AudioFlingerClientAdapter::invalidateStream(audio_stream_type_t stream) {
media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
- return mDelegate->invalidateStream(streamAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->invalidateStream(streamAidl));
}
status_t AudioFlingerClientAdapter::setVoiceVolume(float volume) {
- return mDelegate->setVoiceVolume(volume).transactionError();
+ return statusTFromBinderStatus(mDelegate->setVoiceVolume(volume));
}
status_t AudioFlingerClientAdapter::getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames,
audio_io_handle_t output) const {
int32_t outputAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
media::RenderPosition aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->getRenderPosition(outputAidl, &aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getRenderPosition(outputAidl, &aidlRet)));
if (halFrames != nullptr) {
*halFrames = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(aidlRet.halFrames));
}
@@ -490,7 +492,8 @@
auto result = [&]() -> ConversionResult<uint32_t> {
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->getInputFramesLost(ioHandleAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getInputFramesLost(ioHandleAidl, &aidlRet)));
return convertIntegral<uint32_t>(aidlRet);
}();
// Failure is ignored.
@@ -502,7 +505,8 @@
media::AudioUniqueIdUse useAidl = VALUE_OR_RETURN(
legacy2aidl_audio_unique_id_use_t_AudioUniqueIdUse(use));
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->newAudioUniqueId(useAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->newAudioUniqueId(useAidl, &aidlRet)));
return aidl2legacy_int32_t_audio_unique_id_t(aidlRet);
}();
return result.value_or(AUDIO_UNIQUE_ID_ALLOCATE);
@@ -515,8 +519,8 @@
legacy2aidl_audio_session_t_int32_t(audioSession));
int32_t pidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(pid));
int32_t uidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(uid));
- return mDelegate->acquireAudioSessionId(audioSessionAidl, pidAidl,
- uidAidl).transactionError();
+ return statusTFromBinderStatus(
+ mDelegate->acquireAudioSessionId(audioSessionAidl, pidAidl, uidAidl));
}();
// Failure is ignored.
}
@@ -526,14 +530,16 @@
int32_t audioSessionAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_session_t_int32_t(audioSession));
int32_t pidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(pid));
- return mDelegate->releaseAudioSessionId(audioSessionAidl, pidAidl).transactionError();
+ return statusTFromBinderStatus(
+ mDelegate->releaseAudioSessionId(audioSessionAidl, pidAidl));
}();
// Failure is ignored.
}
status_t AudioFlingerClientAdapter::queryNumberEffects(uint32_t* numEffects) const {
int32_t aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->queryNumberEffects(&aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->queryNumberEffects(&aidlRet)));
if (numEffects != nullptr) {
*numEffects = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(aidlRet));
}
@@ -544,7 +550,8 @@
AudioFlingerClientAdapter::queryEffect(uint32_t index, effect_descriptor_t* pDescriptor) const {
int32_t indexAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(index));
media::EffectDescriptor aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->queryEffect(indexAidl, &aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->queryEffect(indexAidl, &aidlRet)));
if (pDescriptor != nullptr) {
*pDescriptor = VALUE_OR_RETURN_STATUS(
aidl2legacy_EffectDescriptor_effect_descriptor_t(aidlRet));
@@ -563,9 +570,9 @@
int32_t preferredTypeFlagAidl = VALUE_OR_RETURN_STATUS(
convertReinterpret<int32_t>(preferredTypeFlag));
media::EffectDescriptor aidlRet;
- RETURN_STATUS_IF_ERROR(
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
mDelegate->getEffectDescriptor(effectUuidAidl, typeUuidAidl, preferredTypeFlagAidl,
- &aidlRet).transactionError());
+ &aidlRet)));
if (pDescriptor != nullptr) {
*pDescriptor = VALUE_OR_RETURN_STATUS(
aidl2legacy_EffectDescriptor_effect_descriptor_t(aidlRet));
@@ -575,7 +582,7 @@
status_t AudioFlingerClientAdapter::createEffect(const media::CreateEffectRequest& request,
media::CreateEffectResponse* response) {
- return mDelegate->createEffect(request, response).transactionError();
+ return statusTFromBinderStatus(mDelegate->createEffect(request, response));
}
status_t
@@ -586,7 +593,8 @@
legacy2aidl_audio_io_handle_t_int32_t(srcOutput));
int32_t dstOutputAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_io_handle_t_int32_t(dstOutput));
- return mDelegate->moveEffects(sessionAidl, srcOutputAidl, dstOutputAidl).transactionError();
+ return statusTFromBinderStatus(
+ mDelegate->moveEffects(sessionAidl, srcOutputAidl, dstOutputAidl));
}
void AudioFlingerClientAdapter::setEffectSuspended(int effectId,
@@ -596,8 +604,8 @@
int32_t effectIdAidl = VALUE_OR_RETURN_STATUS(convertReinterpret<int32_t>(effectId));
int32_t sessionIdAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_session_t_int32_t(sessionId));
- return mDelegate->setEffectSuspended(effectIdAidl, sessionIdAidl,
- suspended).transactionError();
+ return statusTFromBinderStatus(
+ mDelegate->setEffectSuspended(effectIdAidl, sessionIdAidl, suspended));
}();
// Failure is ignored.
}
@@ -606,7 +614,8 @@
auto result = [&]() -> ConversionResult<audio_module_handle_t> {
std::string nameAidl(name);
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->loadHwModule(nameAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->loadHwModule(nameAidl, &aidlRet)));
return aidl2legacy_int32_t_audio_module_handle_t(aidlRet);
}();
// Failure is ignored.
@@ -616,7 +625,8 @@
uint32_t AudioFlingerClientAdapter::getPrimaryOutputSamplingRate() {
auto result = [&]() -> ConversionResult<uint32_t> {
int32_t aidlRet;
- RETURN_IF_ERROR(mDelegate->getPrimaryOutputSamplingRate(&aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getPrimaryOutputSamplingRate(&aidlRet)));
return convertIntegral<uint32_t>(aidlRet);
}();
// Failure is ignored.
@@ -626,7 +636,8 @@
size_t AudioFlingerClientAdapter::getPrimaryOutputFrameCount() {
auto result = [&]() -> ConversionResult<size_t> {
int64_t aidlRet;
- RETURN_IF_ERROR(mDelegate->getPrimaryOutputFrameCount(&aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getPrimaryOutputFrameCount(&aidlRet)));
return convertIntegral<size_t>(aidlRet);
}();
// Failure is ignored.
@@ -634,13 +645,14 @@
}
status_t AudioFlingerClientAdapter::setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) {
- return mDelegate->setLowRamDevice(isLowRamDevice, totalMemory).transactionError();
+ return statusTFromBinderStatus(mDelegate->setLowRamDevice(isLowRamDevice, totalMemory));
}
status_t AudioFlingerClientAdapter::getAudioPort(struct audio_port_v7* port) {
media::AudioPort portAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_v7_AudioPort(*port));
media::AudioPort aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->getAudioPort(portAidl, &aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getAudioPort(portAidl, &aidlRet)));
*port = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPort_audio_port_v7(aidlRet));
return OK;
}
@@ -649,7 +661,8 @@
audio_patch_handle_t* handle) {
media::AudioPatch patchAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_patch_AudioPatch(*patch));
int32_t aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->createAudioPatch(patchAidl, &aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->createAudioPatch(patchAidl, &aidlRet)));
if (handle != nullptr) {
*handle = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_patch_handle_t(aidlRet));
}
@@ -658,14 +671,15 @@
status_t AudioFlingerClientAdapter::releaseAudioPatch(audio_patch_handle_t handle) {
int32_t handleAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_patch_handle_t_int32_t(handle));
- return mDelegate->releaseAudioPatch(handleAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->releaseAudioPatch(handleAidl));
}
status_t AudioFlingerClientAdapter::listAudioPatches(unsigned int* num_patches,
struct audio_patch* patches) {
std::vector<media::AudioPatch> aidlRet;
int32_t maxPatches = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*num_patches));
- RETURN_STATUS_IF_ERROR(mDelegate->listAudioPatches(maxPatches, &aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->listAudioPatches(maxPatches, &aidlRet)));
*num_patches = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(aidlRet.size()));
return convertRange(aidlRet.begin(), aidlRet.end(), patches,
aidl2legacy_AudioPatch_audio_patch);
@@ -674,29 +688,30 @@
status_t AudioFlingerClientAdapter::setAudioPortConfig(const struct audio_port_config* config) {
media::AudioPortConfig configAidl = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_port_config_AudioPortConfig(*config));
- return mDelegate->setAudioPortConfig(configAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->setAudioPortConfig(configAidl));
}
audio_hw_sync_t AudioFlingerClientAdapter::getAudioHwSyncForSession(audio_session_t sessionId) {
auto result = [&]() -> ConversionResult<audio_hw_sync_t> {
int32_t sessionIdAidl = VALUE_OR_RETURN(legacy2aidl_audio_session_t_int32_t(sessionId));
int32_t aidlRet;
- RETURN_IF_ERROR(
- mDelegate->getAudioHwSyncForSession(sessionIdAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getAudioHwSyncForSession(sessionIdAidl, &aidlRet)));
return aidl2legacy_int32_t_audio_hw_sync_t(aidlRet);
}();
return result.value_or(AUDIO_HW_SYNC_INVALID);
}
status_t AudioFlingerClientAdapter::systemReady() {
- return mDelegate->systemReady().transactionError();
+ return statusTFromBinderStatus(mDelegate->systemReady());
}
size_t AudioFlingerClientAdapter::frameCountHAL(audio_io_handle_t ioHandle) const {
auto result = [&]() -> ConversionResult<size_t> {
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
int64_t aidlRet;
- RETURN_IF_ERROR(mDelegate->frameCountHAL(ioHandleAidl, &aidlRet).transactionError());
+ RETURN_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->frameCountHAL(ioHandleAidl, &aidlRet)));
return convertIntegral<size_t>(aidlRet);
}();
// Failure is ignored.
@@ -706,7 +721,8 @@
status_t
AudioFlingerClientAdapter::getMicrophones(std::vector<media::MicrophoneInfo>* microphones) {
std::vector<media::MicrophoneInfoData> aidlRet;
- RETURN_STATUS_IF_ERROR(mDelegate->getMicrophones(&aidlRet).transactionError());
+ RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
+ mDelegate->getMicrophones(&aidlRet)));
if (microphones != nullptr) {
*microphones = VALUE_OR_RETURN_STATUS(
convertContainer<std::vector<media::MicrophoneInfo>>(aidlRet,
@@ -718,7 +734,7 @@
status_t AudioFlingerClientAdapter::setAudioHalPids(const std::vector<pid_t>& pids) {
std::vector<int32_t> pidsAidl = VALUE_OR_RETURN_STATUS(
convertContainer<std::vector<int32_t>>(pids, legacy2aidl_pid_t_int32_t));
- return mDelegate->setAudioHalPids(pidsAidl).transactionError();
+ return statusTFromBinderStatus(mDelegate->setAudioHalPids(pidsAidl));
}
diff --git a/media/libaudioclient/PlayerBase.cpp b/media/libaudioclient/PlayerBase.cpp
index c443865..9e7d89e 100644
--- a/media/libaudioclient/PlayerBase.cpp
+++ b/media/libaudioclient/PlayerBase.cpp
@@ -15,13 +15,14 @@
*/
#include <binder/IServiceManager.h>
+#include <media/AidlConversionUtil.h>
#include <media/PlayerBase.h>
#define max(a, b) ((a) > (b) ? (a) : (b))
#define min(a, b) ((a) < (b) ? (a) : (b))
namespace android {
-
+using aidl_utils::binderStatusFromStatusT;
using media::VolumeShaperConfiguration;
using media::VolumeShaperOperation;
@@ -150,7 +151,7 @@
if (status != NO_ERROR) {
ALOGW("PlayerBase::setVolume() error %d", status);
}
- return binder::Status::fromStatusT(status);
+ return binderStatusFromStatusT(status);
}
binder::Status PlayerBase::setPan(float pan) {
@@ -170,7 +171,7 @@
if (status != NO_ERROR) {
ALOGW("PlayerBase::setPan() error %d", status);
}
- return binder::Status::fromStatusT(status);
+ return binderStatusFromStatusT(status);
}
binder::Status PlayerBase::setStartDelayMs(int32_t delayMs __unused) {
diff --git a/media/libaudioclient/TrackPlayerBase.cpp b/media/libaudioclient/TrackPlayerBase.cpp
index e571838..372b7c3 100644
--- a/media/libaudioclient/TrackPlayerBase.cpp
+++ b/media/libaudioclient/TrackPlayerBase.cpp
@@ -17,7 +17,7 @@
#include <media/TrackPlayerBase.h>
namespace android {
-
+using aidl_utils::binderStatusFromStatusT;
using media::VolumeShaper;
//--------------------------------------------------------------------------------------------------
@@ -115,7 +115,7 @@
status_t s = spConfiguration->readFromParcelable(configuration)
?: spOperation->readFromParcelable(operation);
if (s != OK) {
- return binder::Status::fromStatusT(s);
+ return binderStatusFromStatusT(s);
}
if (mAudioTrack != 0) {
@@ -124,7 +124,7 @@
if (status < 0) { // a non-negative value is the volume shaper id.
ALOGE("TrackPlayerBase::applyVolumeShaper() failed with status %d", status);
}
- return binder::Status::fromStatusT(status);
+ return binderStatusFromStatusT(status);
} else {
ALOGD("TrackPlayerBase::applyVolumeShaper()"
" no AudioTrack for volume control from IPlayer");
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index a6e5e2e..e3858b9 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -80,10 +80,12 @@
ConversionResult<audio_hw_sync_t> aidl2legacy_int32_t_audio_hw_sync_t(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_hw_sync_t_int32_t(audio_hw_sync_t legacy);
-// The legacy enum is unnamed. Thus, we use int.
-ConversionResult<int> aidl2legacy_AudioPortConfigType(media::AudioPortConfigType aidl);
-// The legacy enum is unnamed. Thus, we use int.
-ConversionResult<media::AudioPortConfigType> legacy2aidl_AudioPortConfigType(int legacy);
+// The legacy enum is unnamed. Thus, we use int32_t.
+ConversionResult<int32_t> aidl2legacy_AudioPortConfigType_int32_t(
+ media::AudioPortConfigType aidl);
+// The legacy enum is unnamed. Thus, we use int32_t.
+ConversionResult<media::AudioPortConfigType> legacy2aidl_int32_t_AudioPortConfigType(
+ int32_t legacy);
ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy);
@@ -129,7 +131,7 @@
legacy2aidl_audio_gain_mode_t_AudioGainMode(audio_gain_mode_t legacy);
ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_mask_int32_t(audio_gain_mode_t legacy);
+ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t_mask(audio_gain_mode_t legacy);
ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl);
ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy);
@@ -149,20 +151,26 @@
ConversionResult<media::AudioOutputFlags> legacy2aidl_audio_output_flags_t_AudioOutputFlags(
audio_output_flags_t legacy);
-ConversionResult<audio_input_flags_t> aidl2legacy_audio_input_flags_mask(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_input_flags_mask(audio_input_flags_t legacy);
+ConversionResult<audio_input_flags_t> aidl2legacy_int32_t_audio_input_flags_t_mask(
+ int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_input_flags_t_int32_t_mask(
+ audio_input_flags_t legacy);
-ConversionResult<audio_output_flags_t> aidl2legacy_audio_output_flags_mask(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_output_flags_mask(audio_output_flags_t legacy);
+ConversionResult<audio_output_flags_t> aidl2legacy_int32_t_audio_output_flags_t_mask(
+ int32_t aidl);
+ConversionResult<int32_t> legacy2aidl_audio_output_flags_t_int32_t_mask(
+ audio_output_flags_t legacy);
ConversionResult<audio_io_flags> aidl2legacy_AudioIoFlags_audio_io_flags(
const media::AudioIoFlags& aidl, media::AudioPortRole role, media::AudioPortType type);
ConversionResult<media::AudioIoFlags> legacy2aidl_audio_io_flags_AudioIoFlags(
const audio_io_flags& legacy, audio_port_role_t role, audio_port_type_t type);
-ConversionResult<audio_port_config_device_ext> aidl2legacy_AudioPortConfigDeviceExt(
+ConversionResult<audio_port_config_device_ext>
+aidl2legacy_AudioPortConfigDeviceExt_audio_port_config_device_ext(
const media::AudioPortConfigDeviceExt& aidl);
-ConversionResult<media::AudioPortConfigDeviceExt> legacy2aidl_AudioPortConfigDeviceExt(
+ConversionResult<media::AudioPortConfigDeviceExt>
+legacy2aidl_audio_port_config_device_ext_AudioPortConfigDeviceExt(
const audio_port_config_device_ext& legacy);
ConversionResult<audio_stream_type_t> aidl2legacy_AudioStreamType_audio_stream_type_t(
@@ -183,9 +191,11 @@
ConversionResult<media::AudioPortConfigMixExt> legacy2aidl_AudioPortConfigMixExt(
const audio_port_config_mix_ext& legacy, audio_port_role_t role);
-ConversionResult<audio_port_config_session_ext> aidl2legacy_AudioPortConfigSessionExt(
+ConversionResult<audio_port_config_session_ext>
+aidl2legacy_AudioPortConfigSessionExt_audio_port_config_session_ext(
const media::AudioPortConfigSessionExt& aidl);
-ConversionResult<media::AudioPortConfigSessionExt> legacy2aidl_AudioPortConfigSessionExt(
+ConversionResult<media::AudioPortConfigSessionExt>
+legacy2aidl_audio_port_config_session_ext_AudioPortConfigSessionExt(
const audio_port_config_session_ext& legacy);
ConversionResult<audio_port_config> aidl2legacy_AudioPortConfig_audio_port_config(
@@ -204,8 +214,10 @@
ConversionResult<media::AudioIoDescriptor> legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(
const sp<AudioIoDescriptor>& legacy);
-ConversionResult<AudioClient> aidl2legacy_AudioClient(const media::AudioClient& aidl);
-ConversionResult<media::AudioClient> legacy2aidl_AudioClient(const AudioClient& legacy);
+ConversionResult<AudioClient> aidl2legacy_AudioClient_AudioClient(
+ const media::AudioClient& aidl);
+ConversionResult<media::AudioClient> legacy2aidl_AudioClient_AudioClient(
+ const AudioClient& legacy);
ConversionResult<audio_content_type_t>
aidl2legacy_AudioContentType_audio_content_type_t(media::AudioContentType aidl);
@@ -263,9 +275,9 @@
legacy2aidl_NullableIMemory_SharedFileRegion(const sp<IMemory>& legacy);
ConversionResult<AudioTimestamp>
-aidl2legacy_AudioTimestamp(const media::AudioTimestampInternal& aidl);
+aidl2legacy_AudioTimestampInternal_AudioTimestamp(const media::AudioTimestampInternal& aidl);
ConversionResult<media::AudioTimestampInternal>
-legacy2aidl_AudioTimestamp(const AudioTimestamp& legacy);
+legacy2aidl_AudioTimestamp_AudioTimestampInternal(const AudioTimestamp& legacy);
ConversionResult<audio_uuid_t>
aidl2legacy_AudioUuid_audio_uuid_t(const media::AudioUuid& aidl);
diff --git a/media/libaudioclient/include/media/AidlConversionUtil.h b/media/libaudioclient/include/media/AidlConversionUtil.h
index 6bfb743..9453673 100644
--- a/media/libaudioclient/include/media/AidlConversionUtil.h
+++ b/media/libaudioclient/include/media/AidlConversionUtil.h
@@ -21,6 +21,7 @@
#include <utility>
#include <android-base/expected.h>
+#include <binder/Status.h>
namespace android {
@@ -132,4 +133,98 @@
#define UNION_SET(u, field, value) \
(u).set<std::decay_t<decltype(u)>::Tag::field>(value)
+namespace aidl_utils {
+
+/**
+ * Return the equivalent Android status_t from a binder exception code.
+ *
+ * Generally one should use statusTFromBinderStatus() instead.
+ *
+ * Exception codes can be generated from a remote Java service exception, translate
+ * them for use on the Native side.
+ *
+ * Note: for EX_TRANSACTION_FAILED and EX_SERVICE_SPECIFIC a more detailed error code
+ * can be found from transactionError() or serviceSpecificErrorCode().
+ */
+static inline status_t statusTFromExceptionCode(int32_t exceptionCode) {
+ using namespace ::android::binder;
+ switch (exceptionCode) {
+ case Status::EX_NONE:
+ return OK;
+ case Status::EX_SECURITY: // Java SecurityException, rethrows locally in Java
+ return PERMISSION_DENIED;
+ case Status::EX_BAD_PARCELABLE: // Java BadParcelableException, rethrows in Java
+ case Status::EX_ILLEGAL_ARGUMENT: // Java IllegalArgumentException, rethrows in Java
+ case Status::EX_NULL_POINTER: // Java NullPointerException, rethrows in Java
+ return BAD_VALUE;
+ case Status::EX_ILLEGAL_STATE: // Java IllegalStateException, rethrows in Java
+ case Status::EX_UNSUPPORTED_OPERATION: // Java UnsupportedOperationException, rethrows
+ return INVALID_OPERATION;
+ case Status::EX_HAS_REPLY_HEADER: // Native strictmode violation
+ case Status::EX_PARCELABLE: // Java bootclass loader (not standard exception), rethrows
+ case Status::EX_NETWORK_MAIN_THREAD: // Java NetworkOnMainThreadException, rethrows
+ case Status::EX_TRANSACTION_FAILED: // Native - see error code
+ case Status::EX_SERVICE_SPECIFIC: // Java ServiceSpecificException,
+ // rethrows in Java with integer error code
+ return UNKNOWN_ERROR;
+ }
+ return UNKNOWN_ERROR;
+}
+
+/**
+ * Return the equivalent Android status_t from a binder status.
+ *
+ * Used to handle errors from a AIDL method declaration
+ *
+ * [oneway] void method(type0 param0, ...)
+ *
+ * or the following (where return_type is not a status_t)
+ *
+ * return_type method(type0 param0, ...)
+ */
+static inline status_t statusTFromBinderStatus(const ::android::binder::Status &status) {
+ return status.isOk() ? OK // check OK,
+ : status.serviceSpecificErrorCode() // service-side error, not standard Java exception
+ // (fromServiceSpecificError)
+ ?: status.transactionError() // a native binder transaction error (fromStatusT)
+ ?: statusTFromExceptionCode(status.exceptionCode()); // a service-side error with a
+ // standard Java exception (fromExceptionCode)
+}
+
+/**
+ * Return a binder::Status from native service status.
+ *
+ * This is used for methods not returning an explicit status_t,
+ * where Java callers expect an exception, not an integer return value.
+ */
+static inline ::android::binder::Status binderStatusFromStatusT(
+ status_t status, const char *optionalMessage = nullptr) {
+ const char * const emptyIfNull = optionalMessage == nullptr ? "" : optionalMessage;
+ // From binder::Status instructions:
+ // Prefer a generic exception code when possible, then a service specific
+ // code, and finally a status_t for low level failures or legacy support.
+ // Exception codes and service specific errors map to nicer exceptions for
+ // Java clients.
+
+ using namespace ::android::binder;
+ switch (status) {
+ case OK:
+ return Status::ok();
+ case PERMISSION_DENIED: // throw SecurityException on Java side
+ return Status::fromExceptionCode(Status::EX_SECURITY, emptyIfNull);
+ case BAD_VALUE: // throw IllegalArgumentException on Java side
+ return Status::fromExceptionCode(Status::EX_ILLEGAL_ARGUMENT, emptyIfNull);
+ case INVALID_OPERATION: // throw IllegalStateException on Java side
+ return Status::fromExceptionCode(Status::EX_ILLEGAL_STATE, emptyIfNull);
+ }
+
+ // A service specific error will not show on status.transactionError() so
+ // be sure to use statusTFromBinderStatus() for reliable error handling.
+
+ // throw a ServiceSpecificException.
+ return Status::fromServiceSpecificError(status, emptyIfNull);
+}
+
+} // namespace aidl_utils
+
} // namespace android
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 350a780..21d18d3 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -7,6 +7,18 @@
}
cc_test {
+ name: "audio_aidl_status_tests",
+ defaults: ["libaudioclient_tests_defaults"],
+ srcs: ["audio_aidl_status_tests.cpp"],
+ shared_libs: [
+ "libaudioclient_aidl_conversion",
+ "libbinder",
+ "libcutils",
+ "libutils",
+ ],
+}
+
+cc_test {
name: "test_create_audiotrack",
defaults: ["libaudioclient_tests_defaults"],
srcs: ["test_create_audiotrack.cpp",
diff --git a/media/libaudioclient/tests/audio_aidl_status_tests.cpp b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
new file mode 100644
index 0000000..5517091
--- /dev/null
+++ b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
@@ -0,0 +1,127 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <gtest/gtest.h>
+#include <media/AidlConversionUtil.h>
+#include <utils/Errors.h>
+
+using namespace android;
+using namespace android::aidl_utils;
+using android::binder::Status;
+
+// Tests for statusTFromBinderStatus() and binderStatusFromStatusT().
+
+// STATUS_T_SMALL_VALUE_LIMIT is an arbitrary limit where we exhaustively check status_t errors.
+// It is known that this limit doesn't cover UNKNOWN_ERROR ~ INT32_MIN.
+constexpr status_t STATUS_T_SMALL_VALUE_LIMIT = -1000;
+
+// Small status values are preserved on round trip
+TEST(audio_aidl_status_tests, statusRoundTripSmallValues) {
+ for (status_t status = 0; status > STATUS_T_SMALL_VALUE_LIMIT; --status) {
+ ASSERT_EQ(status, statusTFromBinderStatus(binderStatusFromStatusT(status)));
+ }
+}
+
+// Special status values are preserved on round trip.
+TEST(audio_aidl_status_tests, statusRoundTripSpecialValues) {
+ for (status_t status : {
+ OK,
+ UNKNOWN_ERROR,
+ NO_MEMORY,
+ INVALID_OPERATION,
+ BAD_VALUE,
+ BAD_TYPE,
+ NAME_NOT_FOUND,
+ PERMISSION_DENIED,
+ NO_INIT,
+ ALREADY_EXISTS,
+ DEAD_OBJECT,
+ FAILED_TRANSACTION,
+ BAD_INDEX,
+ NOT_ENOUGH_DATA,
+ WOULD_BLOCK,
+ TIMED_OUT,
+ UNKNOWN_TRANSACTION,
+ FDS_NOT_ALLOWED}) {
+ ASSERT_EQ(status, statusTFromBinderStatus(binderStatusFromStatusT(status)));
+ }
+}
+
+// Binder exceptions show as an error (not fixed at this time); these come fromExceptionCode().
+TEST(audio_aidl_status_tests, binderStatusExceptions) {
+ for (int exceptionCode : {
+ //Status::EX_NONE,
+ Status::EX_SECURITY,
+ Status::EX_BAD_PARCELABLE,
+ Status::EX_ILLEGAL_ARGUMENT,
+ Status::EX_NULL_POINTER,
+ Status::EX_ILLEGAL_STATE,
+ Status::EX_NETWORK_MAIN_THREAD,
+ Status::EX_UNSUPPORTED_OPERATION,
+ //Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
+ Status::EX_PARCELABLE,
+ // This is special and Java specific; see Parcel.java.
+ Status::EX_HAS_REPLY_HEADER,
+ // This is special, and indicates to C++ binder proxies that the
+ // transaction has failed at a low level.
+ //Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
+ }) {
+ ASSERT_NE(OK, statusTFromBinderStatus(Status::fromExceptionCode(exceptionCode)));
+ }
+}
+
+// Binder transaction errors show exactly in status_t; these come fromStatusT().
+TEST(audio_aidl_status_tests, binderStatusTransactionError) {
+ for (status_t status : {
+ OK, // Note: fromStatusT does check if this is 0, so this is no error.
+ UNKNOWN_ERROR,
+ NO_MEMORY,
+ INVALID_OPERATION,
+ BAD_VALUE,
+ BAD_TYPE,
+ NAME_NOT_FOUND,
+ PERMISSION_DENIED,
+ NO_INIT,
+ ALREADY_EXISTS,
+ DEAD_OBJECT,
+ FAILED_TRANSACTION,
+ BAD_INDEX,
+ NOT_ENOUGH_DATA,
+ WOULD_BLOCK,
+ TIMED_OUT,
+ UNKNOWN_TRANSACTION,
+ FDS_NOT_ALLOWED}) {
+ ASSERT_EQ(status, statusTFromBinderStatus(Status::fromStatusT(status)));
+ }
+}
+
+// Binder service specific errors show in status_t; these come fromServiceSpecificError().
+TEST(audio_aidl_status_tests, binderStatusServiceSpecificError) {
+ // fromServiceSpecificError() still stores exception code if status is 0.
+ for (status_t status = -1; status > STATUS_T_SMALL_VALUE_LIMIT; --status) {
+ ASSERT_EQ(status, statusTFromBinderStatus(Status::fromServiceSpecificError(status)));
+ }
+}
+
+// Binder status with message.
+TEST(audio_aidl_status_tests, binderStatusMessage) {
+ const String8 message("abcd");
+ for (status_t status = -1; status > STATUS_T_SMALL_VALUE_LIMIT; --status) {
+ const Status binderStatus = binderStatusFromStatusT(status, message.c_str());
+ ASSERT_EQ(status, statusTFromBinderStatus(binderStatus));
+ ASSERT_EQ(message, binderStatus.exceptionMessage());
+ }
+}
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index 56343d8..1dee938 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -140,7 +140,7 @@
parcelable->useInChannelMask = mUseInChannelMask;
parcelable->useForVolume = mUseForVolume;
parcelable->mode = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_gain_mode_t_mask_int32_t(mGain.mode));
+ legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
parcelable->channelMask = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
parcelable->minValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_value));
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index 6b63675..20d8632 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -291,7 +291,7 @@
parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
parcelable->gain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.index));
parcelable->gain.mode = VALUE_OR_RETURN_STATUS(
- legacy2aidl_audio_gain_mode_t_mask_int32_t(mGain.mode));
+ legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
parcelable->gain.channelMask = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
parcelable->gain.rampDurationMs = VALUE_OR_RETURN_STATUS(
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index 77c2550..0108816 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -215,7 +215,7 @@
const struct audio_config *config, size_t *size) {
if (mDevice == 0) return NO_INIT;
AudioConfig hidlConfig;
- HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+ HidlUtils::audioConfigFromHal(*config, true /*isInput*/, &hidlConfig);
Result retval;
Return<void> ret = mDevice->getInputBufferSize(
hidlConfig,
@@ -240,7 +240,7 @@
status_t status = deviceAddressFromHal(deviceType, address, &hidlDevice);
if (status != OK) return status;
AudioConfig hidlConfig;
- HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+ HidlUtils::audioConfigFromHal(*config, false /*isInput*/, &hidlConfig);
Result retval = Result::NOT_INITIALIZED;
Return<void> ret = mDevice->openOutputStream(
handle,
@@ -275,7 +275,7 @@
status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
if (status != OK) return status;
AudioConfig hidlConfig;
- HidlUtils::audioConfigFromHal(*config, &hidlConfig);
+ HidlUtils::audioConfigFromHal(*config, true /*isInput*/, &hidlConfig);
Result retval = Result::NOT_INITIALIZED;
#if MAJOR_VERSION == 2
auto sinkMetadata = AudioSource(source);
diff --git a/media/libeffects/lvm/lib/Android.bp b/media/libeffects/lvm/lib/Android.bp
index 8f2f016..dbe0d62 100644
--- a/media/libeffects/lvm/lib/Android.bp
+++ b/media/libeffects/lvm/lib/Android.bp
@@ -131,12 +131,15 @@
shared_libs: [
"liblog",
],
+ static_libs: [
+ "libaudioutils",
+ ],
header_libs: [
"libhardware_headers",
],
cppflags: [
+ "-DBIQUAD_OPT",
"-fvisibility=hidden",
-
"-Wall",
"-Werror",
],
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp b/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
index 5b47aa6..1f0b459 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
@@ -21,6 +21,9 @@
/* */
/****************************************************************************************/
+#ifdef BIQUAD_OPT
+#include <audio_utils/BiquadFilter.h>
+#endif
#include "LVDBE.h"
#include "LVDBE_Private.h"
#include "VectorArithmetic.h"
@@ -107,12 +110,20 @@
/*
* Setup the high pass filter
*/
+#ifdef BIQUAD_OPT
+ std::array<LVM_FLOAT, android::audio_utils::kBiquadNumCoefs> coefs = {
+ LVDBE_HPF_Table[Offset].A0, LVDBE_HPF_Table[Offset].A1, LVDBE_HPF_Table[Offset].A2,
+ -(LVDBE_HPF_Table[Offset].B1), -(LVDBE_HPF_Table[Offset].B2)};
+ pInstance->pBqInstance
+ ->setCoefficients<std::array<LVM_FLOAT, android::audio_utils::kBiquadNumCoefs>>(coefs);
+#else
LoadConst_Float(0, /* Clear the history, value 0 */
(LVM_FLOAT*)&pInstance->pData->HPFTaps, /* Destination */
sizeof(pInstance->pData->HPFTaps) / sizeof(LVM_FLOAT)); /* Number of words */
BQ_2I_D32F32Cll_TRC_WRA_01_Init(&pInstance->pCoef->HPFInstance, /* Initialise the filter */
&pInstance->pData->HPFTaps,
(BQ_FLOAT_Coefs_t*)&LVDBE_HPF_Table[Offset]);
+#endif
/*
* Setup the band pass filter
@@ -275,6 +286,15 @@
LVDBE_Instance_t* pInstance = (LVDBE_Instance_t*)hInstance;
LVMixer3_2St_FLOAT_st* pBypassMixer_Instance = &pInstance->pData->BypassMixer;
+#ifdef BIQUAD_OPT
+ /*
+ * Create biquad instance
+ */
+ pInstance->pBqInstance.reset(
+ new android::audio_utils::BiquadFilter<LVM_FLOAT>(pParams->NrChannels));
+ pInstance->pBqInstance->clear();
+#endif
+
/*
* Update the filters
*/
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
index 12af162..611b762 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
@@ -94,6 +94,14 @@
return LVDBE_NULLADDRESS;
}
+#ifdef BIQUAD_OPT
+ /*
+ * Create biquad instance
+ */
+ pInstance->pBqInstance.reset(
+ new android::audio_utils::BiquadFilter<LVM_FLOAT>(LVM_MAX_CHANNELS));
+#endif
+
/*
* Initialise the filters
*/
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
index 4fef1ef..fa85638 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
@@ -33,6 +33,9 @@
/* */
/****************************************************************************************/
+#ifdef BIQUAD_OPT
+#include <audio_utils/BiquadFilter.h>
+#endif
#include "LVDBE.h" /* Calling or Application layer definitions */
#include "BIQUAD.h"
#include "LVC_Mixer.h"
@@ -63,7 +66,9 @@
AGC_MIX_VOL_2St1Mon_FLOAT_t AGCInstance; /* AGC instance parameters */
/* Process variables */
+#ifndef BIQUAD_OPT
Biquad_2I_Order2_FLOAT_Taps_t HPFTaps; /* High pass filter taps */
+#endif
Biquad_1I_Order2_FLOAT_Taps_t BPFTaps; /* Band pass filter taps */
LVMixer3_1St_FLOAT_st BypassVolume; /* Bypass volume scaler */
LVMixer3_2St_FLOAT_st BypassMixer; /* Bypass Mixer for Click Removal */
@@ -73,7 +78,9 @@
/* Coefs structure */
typedef struct {
/* Process variables */
+#ifndef BIQUAD_OPT
Biquad_FLOAT_Instance_t HPFInstance; /* High pass filter instance */
+#endif
Biquad_FLOAT_Instance_t BPFInstance; /* Band pass filter instance */
} LVDBE_Coef_FLOAT_t;
/* Instance structure */
@@ -86,6 +93,10 @@
LVDBE_Data_FLOAT_t* pData; /* Instance data */
LVDBE_Coef_FLOAT_t* pCoef; /* Instance coefficients */
void* pScratch; /* scratch pointer */
+#ifdef BIQUAD_OPT
+ std::unique_ptr<android::audio_utils::BiquadFilter<LVM_FLOAT>>
+ pBqInstance; /* Biquad filter instance */
+#endif
} LVDBE_Instance_t;
/****************************************************************************************/
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp b/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
index f4a4d6f..bd04a02 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
@@ -20,6 +20,9 @@
/* Includes */
/* */
/****************************************************************************************/
+#ifdef BIQUAD_OPT
+#include <audio_utils/BiquadFilter.h>
+#endif
#include <string.h> // memset
#include "LVDBE.h"
@@ -125,10 +128,14 @@
* Apply the high pass filter if selected
*/
if (pInstance->Params.HPFSelect == LVDBE_HPF_ON) {
+#ifdef BIQUAD_OPT
+ pInstance->pBqInstance->process(pScratch, pScratch, NrFrames);
+#else
BQ_MC_D32F32C30_TRC_WRA_01(&pInstance->pCoef->HPFInstance, /* Filter instance */
pScratch, /* Source */
pScratch, /* Destination */
(LVM_INT16)NrFrames, (LVM_INT16)NrChannels);
+#endif
}
/*
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 670b415..865baad 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -1021,6 +1021,16 @@
ActiveParams.NrChannels = NrChannels;
ActiveParams.ChMask = pConfig->inputCfg.channels;
+ if (NrChannels == 1) {
+ ActiveParams.SourceFormat = LVM_MONO;
+ } else if (NrChannels == 2) {
+ ActiveParams.SourceFormat = LVM_STEREO;
+ } else if (NrChannels > 2 && NrChannels <= LVM_MAX_CHANNELS) {
+ ActiveParams.SourceFormat = LVM_MULTICHANNEL;
+ } else {
+ return -EINVAL;
+ }
+
LvmStatus = LVM_SetControlParameters(pContext->pBundledContext->hInstance, &ActiveParams);
LVM_ERROR_CHECK(LvmStatus, "LVM_SetControlParameters", "Effect_setConfig")
diff --git a/media/libeffects/preprocessing/Android.bp b/media/libeffects/preprocessing/Android.bp
index 5217cf9..681e247 100644
--- a/media/libeffects/preprocessing/Android.bp
+++ b/media/libeffects/preprocessing/Android.bp
@@ -1,35 +1,5 @@
// audio preprocessing wrapper
cc_library_shared {
- name: "libaudiopreprocessing_legacy",
-
- vendor: true,
-
- relative_install_path: "soundfx",
-
- srcs: ["PreProcessing.cpp"],
-
- shared_libs: [
- "libwebrtc_audio_preprocessing",
- "libspeexresampler",
- "libutils",
- "liblog",
- ],
-
- cflags: [
- "-DWEBRTC_POSIX",
- "-DWEBRTC_LEGACY",
- "-fvisibility=hidden",
- "-Wall",
- "-Werror",
- ],
-
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
-}
-
-cc_library_shared {
name: "libaudiopreprocessing",
vendor: true,
relative_install_path: "soundfx",
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 1a5547b..03ccc34 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -23,15 +23,10 @@
#include <hardware/audio_effect.h>
#include <utils/Log.h>
#include <utils/Timers.h>
-#ifndef WEBRTC_LEGACY
#include <audio_effects/effect_agc2.h>
-#endif
#include <audio_effects/effect_ns.h>
#include <audio_processing.h>
#include <module_common_types.h>
-#ifdef WEBRTC_LEGACY
-#include "speex/speex_resampler.h"
-#endif
// undefine to perform multi channels API functional tests
//#define DUAL_MIC_TEST
@@ -46,9 +41,7 @@
// types of pre processing modules
enum preproc_id {
PREPROC_AGC, // Automatic Gain Control
-#ifndef WEBRTC_LEGACY
PREPROC_AGC2, // Automatic Gain Control 2
-#endif
PREPROC_AEC, // Acoustic Echo Canceler
PREPROC_NS, // Noise Suppressor
PREPROC_NUM_EFFECTS
@@ -110,10 +103,8 @@
int id; // audio session ID
int io; // handle of input stream this session is on
webrtc::AudioProcessing* apm; // handle on webRTC audio processing module (APM)
-#ifndef WEBRTC_LEGACY
// Audio Processing module builder
webrtc::AudioProcessingBuilder ap_builder;
-#endif
size_t apmFrameCount; // buffer size for webRTC process (10 ms)
uint32_t apmSamplingRate; // webRTC APM sampling rate (8/16 or 32 kHz)
size_t frameCount; // buffer size before input resampler ( <=> apmFrameCount)
@@ -124,42 +115,25 @@
uint32_t enabledMsk; // bit field containing IDs of enabled pre processors
uint32_t processedMsk; // bit field containing IDs of pre processors already
// processed in current round
-#ifdef WEBRTC_LEGACY
- webrtc::AudioFrame* procFrame; // audio frame passed to webRTC AMP ProcessStream()
-#else
// audio config strucutre
webrtc::AudioProcessing::Config config;
webrtc::StreamConfig inputConfig; // input stream configuration
webrtc::StreamConfig outputConfig; // output stream configuration
-#endif
int16_t* inBuf; // input buffer used when resampling
size_t inBufSize; // input buffer size in frames
size_t framesIn; // number of frames in input buffer
-#ifdef WEBRTC_LEGACY
- SpeexResamplerState* inResampler; // handle on input speex resampler
-#endif
int16_t* outBuf; // output buffer used when resampling
size_t outBufSize; // output buffer size in frames
size_t framesOut; // number of frames in output buffer
-#ifdef WEBRTC_LEGACY
- SpeexResamplerState* outResampler; // handle on output speex resampler
-#endif
uint32_t revChannelCount; // number of channels on reverse stream
uint32_t revEnabledMsk; // bit field containing IDs of enabled pre processors
// with reverse channel
uint32_t revProcessedMsk; // bit field containing IDs of pre processors with reverse
// channel already processed in current round
-#ifdef WEBRTC_LEGACY
- webrtc::AudioFrame* revFrame; // audio frame passed to webRTC AMP AnalyzeReverseStream()
-#else
webrtc::StreamConfig revConfig; // reverse stream configuration.
-#endif
int16_t* revBuf; // reverse channel input buffer
size_t revBufSize; // reverse channel input buffer size
size_t framesRev; // number of frames in reverse channel input buffer
-#ifdef WEBRTC_LEGACY
- SpeexResamplerState* revResampler; // handle on reverse channel input speex resampler
-#endif
};
#ifdef DUAL_MIC_TEST
@@ -213,7 +187,6 @@
"Automatic Gain Control",
"The Android Open Source Project"};
-#ifndef WEBRTC_LEGACY
// Automatic Gain Control 2
static const effect_descriptor_t sAgc2Descriptor = {
{0xae3c653b, 0xbe18, 0x4ab8, 0x8938, {0x41, 0x8f, 0x0a, 0x7f, 0x06, 0xac}}, // type
@@ -224,7 +197,6 @@
0, // FIXME indicate memory usage
"Automatic Gain Control 2",
"The Android Open Source Project"};
-#endif
// Acoustic Echo Cancellation
static const effect_descriptor_t sAecDescriptor = {
@@ -249,9 +221,7 @@
"The Android Open Source Project"};
static const effect_descriptor_t* sDescriptors[PREPROC_NUM_EFFECTS] = {&sAgcDescriptor,
-#ifndef WEBRTC_LEGACY
&sAgc2Descriptor,
-#endif
&sAecDescriptor,
&sNsDescriptor};
@@ -260,9 +230,7 @@
//------------------------------------------------------------------------------
const effect_uuid_t* const sUuidToPreProcTable[PREPROC_NUM_EFFECTS] = {FX_IID_AGC,
-#ifndef WEBRTC_LEGACY
FX_IID_AGC2,
-#endif
FX_IID_AEC, FX_IID_NS};
const effect_uuid_t* ProcIdToUuid(int procId) {
@@ -297,7 +265,6 @@
static const int kAgcDefaultCompGain = 9;
static const bool kAgcDefaultLimiter = true;
-#ifndef WEBRTC_LEGACY
int Agc2Init(preproc_effect_t* effect) {
ALOGV("Agc2Init");
effect->session->config = effect->session->apm->GetConfig();
@@ -308,48 +275,27 @@
effect->session->apm->ApplyConfig(effect->session->config);
return 0;
}
-#endif
int AgcInit(preproc_effect_t* effect) {
ALOGV("AgcInit");
-#ifdef WEBRTC_LEGACY
- webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
- agc->set_mode(webrtc::GainControl::kFixedDigital);
- agc->set_target_level_dbfs(kAgcDefaultTargetLevel);
- agc->set_compression_gain_db(kAgcDefaultCompGain);
- agc->enable_limiter(kAgcDefaultLimiter);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller1.target_level_dbfs = kAgcDefaultTargetLevel;
effect->session->config.gain_controller1.compression_gain_db = kAgcDefaultCompGain;
effect->session->config.gain_controller1.enable_limiter = kAgcDefaultLimiter;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
return 0;
}
-#ifndef WEBRTC_LEGACY
int Agc2Create(preproc_effect_t* effect) {
Agc2Init(effect);
return 0;
}
-#endif
int AgcCreate(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::GainControl* agc = effect->session->apm->gain_control();
- ALOGV("AgcCreate got agc %p", agc);
- if (agc == NULL) {
- ALOGW("AgcCreate Error");
- return -ENOMEM;
- }
- effect->engine = static_cast<preproc_fx_handle_t>(agc);
-#endif
AgcInit(effect);
return 0;
}
-#ifndef WEBRTC_LEGACY
int Agc2GetParameter(preproc_effect_t* effect, void* pParam, uint32_t* pValueSize, void* pValue) {
int status = 0;
uint32_t param = *(uint32_t*)pParam;
@@ -422,15 +368,11 @@
return status;
}
-#endif
int AgcGetParameter(preproc_effect_t* effect, void* pParam, uint32_t* pValueSize, void* pValue) {
int status = 0;
uint32_t param = *(uint32_t*)pParam;
t_agc_settings* pProperties = (t_agc_settings*)pValue;
-#ifdef WEBRTC_LEGACY
- webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
-#endif
switch (param) {
case AGC_PARAM_TARGET_LEVEL:
@@ -459,32 +401,6 @@
break;
}
-#ifdef WEBRTC_LEGACY
- switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- *(int16_t*)pValue = (int16_t)(agc->target_level_dbfs() * -100);
- ALOGV("AgcGetParameter() target level %d milliBels", *(int16_t*)pValue);
- break;
- case AGC_PARAM_COMP_GAIN:
- *(int16_t*)pValue = (int16_t)(agc->compression_gain_db() * 100);
- ALOGV("AgcGetParameter() comp gain %d milliBels", *(int16_t*)pValue);
- break;
- case AGC_PARAM_LIMITER_ENA:
- *(bool*)pValue = (bool)agc->is_limiter_enabled();
- ALOGV("AgcGetParameter() limiter enabled %s",
- (*(int16_t*)pValue != 0) ? "true" : "false");
- break;
- case AGC_PARAM_PROPERTIES:
- pProperties->targetLevel = (int16_t)(agc->target_level_dbfs() * -100);
- pProperties->compGain = (int16_t)(agc->compression_gain_db() * 100);
- pProperties->limiterEnabled = (bool)agc->is_limiter_enabled();
- break;
- default:
- ALOGW("AgcGetParameter() unknown param %d", param);
- status = -EINVAL;
- break;
- }
-#else
effect->session->config = effect->session->apm->GetConfig();
switch (param) {
case AGC_PARAM_TARGET_LEVEL:
@@ -515,11 +431,9 @@
status = -EINVAL;
break;
}
-#endif
return status;
}
-#ifndef WEBRTC_LEGACY
int Agc2SetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
uint32_t param = *(uint32_t*)pParam;
@@ -567,43 +481,9 @@
return status;
}
-#endif
int AgcSetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
-#ifdef WEBRTC_LEGACY
- uint32_t param = *(uint32_t*)pParam;
- t_agc_settings* pProperties = (t_agc_settings*)pValue;
- webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
-
- switch (param) {
- case AGC_PARAM_TARGET_LEVEL:
- ALOGV("AgcSetParameter() target level %d milliBels", *(int16_t*)pValue);
- status = agc->set_target_level_dbfs(-(*(int16_t*)pValue / 100));
- break;
- case AGC_PARAM_COMP_GAIN:
- ALOGV("AgcSetParameter() comp gain %d milliBels", *(int16_t*)pValue);
- status = agc->set_compression_gain_db(*(int16_t*)pValue / 100);
- break;
- case AGC_PARAM_LIMITER_ENA:
- ALOGV("AgcSetParameter() limiter enabled %s", *(bool*)pValue ? "true" : "false");
- status = agc->enable_limiter(*(bool*)pValue);
- break;
- case AGC_PARAM_PROPERTIES:
- ALOGV("AgcSetParameter() properties level %d, gain %d limiter %d",
- pProperties->targetLevel, pProperties->compGain, pProperties->limiterEnabled);
- status = agc->set_target_level_dbfs(-(pProperties->targetLevel / 100));
- if (status != 0) break;
- status = agc->set_compression_gain_db(pProperties->compGain / 100);
- if (status != 0) break;
- status = agc->enable_limiter(pProperties->limiterEnabled);
- break;
- default:
- ALOGW("AgcSetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
- status = -EINVAL;
- break;
- }
-#else
uint32_t param = *(uint32_t*)pParam;
t_agc_settings* pProperties = (t_agc_settings*)pValue;
effect->session->config = effect->session->apm->GetConfig();
@@ -637,96 +517,57 @@
break;
}
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
ALOGV("AgcSetParameter() done status %d", status);
return status;
}
-#ifndef WEBRTC_LEGACY
void Agc2Enable(preproc_effect_t* effect) {
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller2.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
}
-#endif
void AgcEnable(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
- ALOGV("AgcEnable agc %p", agc);
- agc->Enable(true);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller1.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
-#ifndef WEBRTC_LEGACY
void Agc2Disable(preproc_effect_t* effect) {
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller2.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
}
-#endif
void AgcDisable(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- ALOGV("AgcDisable");
- webrtc::GainControl* agc = static_cast<webrtc::GainControl*>(effect->engine);
- agc->Enable(false);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.gain_controller1.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
static const preproc_ops_t sAgcOps = {AgcCreate, AgcInit, NULL, AgcEnable, AgcDisable,
AgcSetParameter, AgcGetParameter, NULL};
-#ifndef WEBRTC_LEGACY
static const preproc_ops_t sAgc2Ops = {Agc2Create, Agc2Init, NULL,
Agc2Enable, Agc2Disable, Agc2SetParameter,
Agc2GetParameter, NULL};
-#endif
//------------------------------------------------------------------------------
// Acoustic Echo Canceler (AEC)
//------------------------------------------------------------------------------
-#ifdef WEBRTC_LEGACY
-static const webrtc::EchoControlMobile::RoutingMode kAecDefaultMode =
- webrtc::EchoControlMobile::kEarpiece;
-static const bool kAecDefaultComfortNoise = true;
-#endif
int AecInit(preproc_effect_t* effect) {
ALOGV("AecInit");
-#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
- aec->set_routing_mode(kAecDefaultMode);
- aec->enable_comfort_noise(kAecDefaultComfortNoise);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.echo_canceller.mobile_mode = true;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
return 0;
}
int AecCreate(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile* aec = effect->session->apm->echo_control_mobile();
- ALOGV("AecCreate got aec %p", aec);
- if (aec == NULL) {
- ALOGW("AgcCreate Error");
- return -ENOMEM;
- }
- effect->engine = static_cast<preproc_fx_handle_t>(aec);
-#endif
AecInit(effect);
return 0;
}
@@ -744,13 +585,11 @@
*(uint32_t*)pValue = 1000 * effect->session->apm->stream_delay_ms();
ALOGV("AecGetParameter() echo delay %d us", *(uint32_t*)pValue);
break;
-#ifndef WEBRTC_LEGACY
case AEC_PARAM_MOBILE_MODE:
effect->session->config = effect->session->apm->GetConfig();
*(uint32_t*)pValue = effect->session->config.echo_canceller.mobile_mode;
ALOGV("AecGetParameter() mobile mode %d us", *(uint32_t*)pValue);
break;
-#endif
default:
ALOGW("AecGetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
status = -EINVAL;
@@ -770,14 +609,12 @@
status = effect->session->apm->set_stream_delay_ms(value / 1000);
ALOGV("AecSetParameter() echo delay %d us, status %d", value, status);
break;
-#ifndef WEBRTC_LEGACY
case AEC_PARAM_MOBILE_MODE:
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.echo_canceller.mobile_mode = value;
ALOGV("AecSetParameter() mobile mode %d us", value);
effect->session->apm->ApplyConfig(effect->session->config);
break;
-#endif
default:
ALOGW("AecSetParameter() unknown param %08x value %08x", param, *(uint32_t*)pValue);
status = -EINVAL;
@@ -787,57 +624,24 @@
}
void AecEnable(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
- ALOGV("AecEnable aec %p", aec);
- aec->Enable(true);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.echo_canceller.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
void AecDisable(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- ALOGV("AecDisable");
- webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
- aec->Enable(false);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.echo_canceller.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
int AecSetDevice(preproc_effect_t* effect, uint32_t device) {
ALOGV("AecSetDevice %08x", device);
-#ifdef WEBRTC_LEGACY
- webrtc::EchoControlMobile* aec = static_cast<webrtc::EchoControlMobile*>(effect->engine);
- webrtc::EchoControlMobile::RoutingMode mode =
- webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
-#endif
if (audio_is_input_device(device)) {
return 0;
}
-#ifdef WEBRTC_LEGACY
- switch (device) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- mode = webrtc::EchoControlMobile::kEarpiece;
- break;
- case AUDIO_DEVICE_OUT_SPEAKER:
- mode = webrtc::EchoControlMobile::kSpeakerphone;
- break;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_USB_HEADSET:
- default:
- break;
- }
- aec->set_routing_mode(mode);
-#endif
return 0;
}
@@ -849,49 +653,19 @@
// Noise Suppression (NS)
//------------------------------------------------------------------------------
-#ifdef WEBRTC_LEGACY
-static const webrtc::NoiseSuppression::Level kNsDefaultLevel = webrtc::NoiseSuppression::kModerate;
-#else
static const webrtc::AudioProcessing::Config::NoiseSuppression::Level kNsDefaultLevel =
webrtc::AudioProcessing::Config::NoiseSuppression::kModerate;
-#endif
int NsInit(preproc_effect_t* effect) {
ALOGV("NsInit");
-#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
- ns->set_level(kNsDefaultLevel);
- webrtc::Config config;
- std::vector<webrtc::Point> geometry;
- // TODO(aluebs): Make the geometry settable.
- geometry.push_back(webrtc::Point(-0.03f, 0.f, 0.f));
- geometry.push_back(webrtc::Point(-0.01f, 0.f, 0.f));
- geometry.push_back(webrtc::Point(0.01f, 0.f, 0.f));
- geometry.push_back(webrtc::Point(0.03f, 0.f, 0.f));
- // The geometry needs to be set with Beamforming enabled.
- config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
- effect->session->apm->SetExtraOptions(config);
- config.Set<webrtc::Beamforming>(new webrtc::Beamforming(false, geometry));
- effect->session->apm->SetExtraOptions(config);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.noise_suppression.level = kNsDefaultLevel;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
effect->type = NS_TYPE_SINGLE_CHANNEL;
return 0;
}
int NsCreate(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression* ns = effect->session->apm->noise_suppression();
- ALOGV("NsCreate got ns %p", ns);
- if (ns == NULL) {
- ALOGW("AgcCreate Error");
- return -ENOMEM;
- }
- effect->engine = static_cast<preproc_fx_handle_t>(ns);
-#endif
NsInit(effect);
return 0;
}
@@ -904,31 +678,6 @@
int NsSetParameter(preproc_effect_t* effect, void* pParam, void* pValue) {
int status = 0;
-#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
- uint32_t param = *(uint32_t*)pParam;
- uint32_t value = *(uint32_t*)pValue;
- switch (param) {
- case NS_PARAM_LEVEL:
- ns->set_level((webrtc::NoiseSuppression::Level)value);
- ALOGV("NsSetParameter() level %d", value);
- break;
- case NS_PARAM_TYPE: {
- webrtc::Config config;
- std::vector<webrtc::Point> geometry;
- bool is_beamforming_enabled = value == NS_TYPE_MULTI_CHANNEL && ns->is_enabled();
- config.Set<webrtc::Beamforming>(
- new webrtc::Beamforming(is_beamforming_enabled, geometry));
- effect->session->apm->SetExtraOptions(config);
- effect->type = value;
- ALOGV("NsSetParameter() type %d", value);
- break;
- }
- default:
- ALOGW("NsSetParameter() unknown param %08x value %08x", param, value);
- status = -EINVAL;
- }
-#else
uint32_t param = *(uint32_t*)pParam;
uint32_t value = *(uint32_t*)pValue;
effect->session->config = effect->session->apm->GetConfig();
@@ -943,52 +692,28 @@
status = -EINVAL;
}
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
return status;
}
void NsEnable(preproc_effect_t* effect) {
-#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
- ALOGV("NsEnable ns %p", ns);
- ns->Enable(true);
- if (effect->type == NS_TYPE_MULTI_CHANNEL) {
- webrtc::Config config;
- std::vector<webrtc::Point> geometry;
- config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
- effect->session->apm->SetExtraOptions(config);
- }
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.noise_suppression.enabled = true;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
void NsDisable(preproc_effect_t* effect) {
ALOGV("NsDisable");
-#ifdef WEBRTC_LEGACY
- webrtc::NoiseSuppression* ns = static_cast<webrtc::NoiseSuppression*>(effect->engine);
- ns->Enable(false);
- webrtc::Config config;
- std::vector<webrtc::Point> geometry;
- config.Set<webrtc::Beamforming>(new webrtc::Beamforming(false, geometry));
- effect->session->apm->SetExtraOptions(config);
-#else
effect->session->config = effect->session->apm->GetConfig();
effect->session->config.noise_suppression.enabled = false;
effect->session->apm->ApplyConfig(effect->session->config);
-#endif
}
static const preproc_ops_t sNsOps = {NsCreate, NsInit, NULL, NsEnable,
NsDisable, NsSetParameter, NsGetParameter, NULL};
static const preproc_ops_t* sPreProcOps[PREPROC_NUM_EFFECTS] = {&sAgcOps,
-#ifndef WEBRTC_LEGACY
&sAgc2Ops,
-#endif
&sAecOps, &sNsOps};
//------------------------------------------------------------------------------
@@ -1119,9 +844,6 @@
session->id = 0;
session->io = 0;
session->createdMsk = 0;
-#ifdef WEBRTC_LEGACY
- session->apm = NULL;
-#endif
for (i = 0; i < PREPROC_NUM_EFFECTS && status == 0; i++) {
status = Effect_Init(&session->effects[i], i);
}
@@ -1135,75 +857,32 @@
ALOGV("Session_CreateEffect procId %d, createdMsk %08x", procId, session->createdMsk);
if (session->createdMsk == 0) {
-#ifdef WEBRTC_LEGACY
- session->apm = webrtc::AudioProcessing::Create();
- if (session->apm == NULL) {
- ALOGW("Session_CreateEffect could not get apm engine");
- goto error;
- }
- const webrtc::ProcessingConfig processing_config = {
- {{kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl},
- {kPreprocDefaultSr, kPreProcDefaultCnl}}};
- session->apm->Initialize(processing_config);
- session->procFrame = new webrtc::AudioFrame();
- if (session->procFrame == NULL) {
- ALOGW("Session_CreateEffect could not allocate audio frame");
- goto error;
- }
- session->revFrame = new webrtc::AudioFrame();
- if (session->revFrame == NULL) {
- ALOGW("Session_CreateEffect could not allocate reverse audio frame");
- goto error;
- }
-#else
session->apm = session->ap_builder.Create();
if (session->apm == NULL) {
ALOGW("Session_CreateEffect could not get apm engine");
goto error;
}
-#endif
session->apmSamplingRate = kPreprocDefaultSr;
session->apmFrameCount = (kPreprocDefaultSr) / 100;
session->frameCount = session->apmFrameCount;
session->samplingRate = kPreprocDefaultSr;
session->inChannelCount = kPreProcDefaultCnl;
session->outChannelCount = kPreProcDefaultCnl;
-#ifdef WEBRTC_LEGACY
- session->procFrame->sample_rate_hz_ = kPreprocDefaultSr;
- session->procFrame->num_channels_ = kPreProcDefaultCnl;
-#else
session->inputConfig.set_sample_rate_hz(kPreprocDefaultSr);
session->inputConfig.set_num_channels(kPreProcDefaultCnl);
session->outputConfig.set_sample_rate_hz(kPreprocDefaultSr);
session->outputConfig.set_num_channels(kPreProcDefaultCnl);
-#endif
session->revChannelCount = kPreProcDefaultCnl;
-#ifdef WEBRTC_LEGACY
- session->revFrame->sample_rate_hz_ = kPreprocDefaultSr;
- session->revFrame->num_channels_ = kPreProcDefaultCnl;
-#else
session->revConfig.set_sample_rate_hz(kPreprocDefaultSr);
session->revConfig.set_num_channels(kPreProcDefaultCnl);
-#endif
session->enabledMsk = 0;
session->processedMsk = 0;
session->revEnabledMsk = 0;
session->revProcessedMsk = 0;
-#ifdef WEBRTC_LEGACY
- session->inResampler = NULL;
-#endif
session->inBuf = NULL;
session->inBufSize = 0;
-#ifdef WEBRTC_LEGACY
- session->outResampler = NULL;
-#endif
session->outBuf = NULL;
session->outBufSize = 0;
-#ifdef WEBRTC_LEGACY
- session->revResampler = NULL;
-#endif
session->revBuf = NULL;
session->revBufSize = 0;
}
@@ -1217,17 +896,8 @@
error:
if (session->createdMsk == 0) {
-#ifdef WEBRTC_LEGACY
- delete session->revFrame;
- session->revFrame = NULL;
- delete session->procFrame;
- session->procFrame = NULL;
- delete session->apm;
- session->apm = NULL; // NOLINT(clang-analyzer-cplusplus.NewDelete)
-#else
delete session->apm;
session->apm = NULL;
-#endif
}
return status;
}
@@ -1236,29 +906,8 @@
ALOGW_IF(Effect_Release(fx) != 0, " Effect_Release() failed for proc ID %d", fx->procId);
session->createdMsk &= ~(1 << fx->procId);
if (session->createdMsk == 0) {
-#ifdef WEBRTC_LEGACY
delete session->apm;
session->apm = NULL;
- delete session->procFrame;
- session->procFrame = NULL;
- delete session->revFrame;
- session->revFrame = NULL;
- if (session->inResampler != NULL) {
- speex_resampler_destroy(session->inResampler);
- session->inResampler = NULL;
- }
- if (session->outResampler != NULL) {
- speex_resampler_destroy(session->outResampler);
- session->outResampler = NULL;
- }
- if (session->revResampler != NULL) {
- speex_resampler_destroy(session->revResampler);
- session->revResampler = NULL;
- }
-#else
- delete session->apm;
- session->apm = NULL;
-#endif
delete session->inBuf;
session->inBuf = NULL;
delete session->outBuf;
@@ -1284,9 +933,6 @@
ALOGV("Session_SetConfig sr %d cnl %08x", config->inputCfg.samplingRate,
config->inputCfg.channels);
-#ifdef WEBRTC_LEGACY
- int status;
-#endif
// AEC implementation is limited to 16kHz
if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
@@ -1297,51 +943,25 @@
session->apmSamplingRate = 8000;
}
-#ifdef WEBRTC_LEGACY
- const webrtc::ProcessingConfig processing_config = {
- {{static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), outCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl}}};
- status = session->apm->Initialize(processing_config);
- if (status < 0) {
- return -EINVAL;
- }
-#endif
session->samplingRate = config->inputCfg.samplingRate;
session->apmFrameCount = session->apmSamplingRate / 100;
if (session->samplingRate == session->apmSamplingRate) {
session->frameCount = session->apmFrameCount;
} else {
-#ifdef WEBRTC_LEGACY
- session->frameCount =
- (session->apmFrameCount * session->samplingRate) / session->apmSamplingRate + 1;
-#else
session->frameCount =
(session->apmFrameCount * session->samplingRate) / session->apmSamplingRate;
-#endif
}
session->inChannelCount = inCnl;
session->outChannelCount = outCnl;
-#ifdef WEBRTC_LEGACY
- session->procFrame->num_channels_ = inCnl;
- session->procFrame->sample_rate_hz_ = session->apmSamplingRate;
-#else
session->inputConfig.set_sample_rate_hz(session->samplingRate);
session->inputConfig.set_num_channels(inCnl);
session->outputConfig.set_sample_rate_hz(session->samplingRate);
session->outputConfig.set_num_channels(inCnl);
-#endif
session->revChannelCount = inCnl;
-#ifdef WEBRTC_LEGACY
- session->revFrame->num_channels_ = inCnl;
- session->revFrame->sample_rate_hz_ = session->apmSamplingRate;
-#else
session->revConfig.set_sample_rate_hz(session->samplingRate);
session->revConfig.set_num_channels(inCnl);
-#endif
// force process buffer reallocation
session->inBufSize = 0;
@@ -1349,53 +969,6 @@
session->framesIn = 0;
session->framesOut = 0;
-#ifdef WEBRTC_LEGACY
- if (session->inResampler != NULL) {
- speex_resampler_destroy(session->inResampler);
- session->inResampler = NULL;
- }
- if (session->outResampler != NULL) {
- speex_resampler_destroy(session->outResampler);
- session->outResampler = NULL;
- }
- if (session->revResampler != NULL) {
- speex_resampler_destroy(session->revResampler);
- session->revResampler = NULL;
- }
- if (session->samplingRate != session->apmSamplingRate) {
- int error;
- session->inResampler =
- speex_resampler_init(session->inChannelCount, session->samplingRate,
- session->apmSamplingRate, RESAMPLER_QUALITY, &error);
- if (session->inResampler == NULL) {
- ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
- return -EINVAL;
- }
- session->outResampler =
- speex_resampler_init(session->outChannelCount, session->apmSamplingRate,
- session->samplingRate, RESAMPLER_QUALITY, &error);
- if (session->outResampler == NULL) {
- ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
- speex_resampler_destroy(session->inResampler);
- session->inResampler = NULL;
- return -EINVAL;
- }
- session->revResampler =
- speex_resampler_init(session->inChannelCount, session->samplingRate,
- session->apmSamplingRate, RESAMPLER_QUALITY, &error);
- if (session->revResampler == NULL) {
- ALOGW("Session_SetConfig Cannot create speex resampler: %s",
- speex_resampler_strerror(error));
- speex_resampler_destroy(session->inResampler);
- session->inResampler = NULL;
- speex_resampler_destroy(session->outResampler);
- session->outResampler = NULL;
- return -EINVAL;
- }
- }
-#endif
session->state = PREPROC_SESSION_STATE_CONFIG;
return 0;
@@ -1430,22 +1003,7 @@
return -EINVAL;
}
uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
-#ifdef WEBRTC_LEGACY
- const webrtc::ProcessingConfig processing_config = {
- {{static_cast<int>(session->apmSamplingRate), session->inChannelCount},
- {static_cast<int>(session->apmSamplingRate), session->outChannelCount},
- {static_cast<int>(session->apmSamplingRate), inCnl},
- {static_cast<int>(session->apmSamplingRate), inCnl}}};
- int status = session->apm->Initialize(processing_config);
- if (status < 0) {
- return -EINVAL;
- }
-#endif
session->revChannelCount = inCnl;
-#ifdef WEBRTC_LEGACY
- session->revFrame->num_channels_ = inCnl;
- session->revFrame->sample_rate_hz_ = session->apmSamplingRate;
-#endif
// force process buffer reallocation
session->revBufSize = 0;
session->framesRev = 0;
@@ -1467,24 +1025,10 @@
if (enabled) {
if (session->enabledMsk == 0) {
session->framesIn = 0;
-#ifdef WEBRTC_LEGACY
- if (session->inResampler != NULL) {
- speex_resampler_reset_mem(session->inResampler);
- }
- session->framesOut = 0;
- if (session->outResampler != NULL) {
- speex_resampler_reset_mem(session->outResampler);
- }
-#endif
}
session->enabledMsk |= (1 << procId);
if (HasReverseStream(procId)) {
session->framesRev = 0;
-#ifdef WEBRTC_LEGACY
- if (session->revResampler != NULL) {
- speex_resampler_reset_mem(session->revResampler);
- }
-#endif
session->revEnabledMsk |= (1 << procId);
}
} else {
@@ -1600,82 +1144,6 @@
return 0;
}
-#ifdef WEBRTC_LEGACY
- if (session->inResampler != NULL) {
- size_t fr = session->frameCount - session->framesIn;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- if (session->inBufSize < session->framesIn + fr) {
- int16_t* buf;
- session->inBufSize = session->framesIn + fr;
- buf = (int16_t*)realloc(
- session->inBuf,
- session->inBufSize * session->inChannelCount * sizeof(int16_t));
- if (buf == NULL) {
- session->framesIn = 0;
- free(session->inBuf);
- session->inBuf = NULL;
- return -ENOMEM;
- }
- session->inBuf = buf;
- }
- memcpy(session->inBuf + session->framesIn * session->inChannelCount, inBuffer->s16,
- fr * session->inChannelCount * sizeof(int16_t));
-#ifdef DUAL_MIC_TEST
- pthread_mutex_lock(&gPcmDumpLock);
- if (gPcmDumpFh != NULL) {
- fwrite(inBuffer->raw, fr * session->inChannelCount * sizeof(int16_t), 1,
- gPcmDumpFh);
- }
- pthread_mutex_unlock(&gPcmDumpLock);
-#endif
-
- session->framesIn += fr;
- inBuffer->frameCount = fr;
- if (session->framesIn < session->frameCount) {
- return 0;
- }
- spx_uint32_t frIn = session->framesIn;
- spx_uint32_t frOut = session->apmFrameCount;
- if (session->inChannelCount == 1) {
- speex_resampler_process_int(session->inResampler, 0, session->inBuf, &frIn,
- session->procFrame->data_, &frOut);
- } else {
- speex_resampler_process_interleaved_int(session->inResampler, session->inBuf, &frIn,
- session->procFrame->data_, &frOut);
- }
- memmove(session->inBuf, session->inBuf + frIn * session->inChannelCount,
- (session->framesIn - frIn) * session->inChannelCount * sizeof(int16_t));
- session->framesIn -= frIn;
- } else {
- size_t fr = session->frameCount - session->framesIn;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- memcpy(session->procFrame->data_ + session->framesIn * session->inChannelCount,
- inBuffer->s16, fr * session->inChannelCount * sizeof(int16_t));
-
-#ifdef DUAL_MIC_TEST
- pthread_mutex_lock(&gPcmDumpLock);
- if (gPcmDumpFh != NULL) {
- fwrite(inBuffer->raw, fr * session->inChannelCount * sizeof(int16_t), 1,
- gPcmDumpFh);
- }
- pthread_mutex_unlock(&gPcmDumpLock);
-#endif
-
- session->framesIn += fr;
- inBuffer->frameCount = fr;
- if (session->framesIn < session->frameCount) {
- return 0;
- }
- session->framesIn = 0;
- }
- session->procFrame->samples_per_channel_ = session->apmFrameCount;
-
- effect->session->apm->ProcessStream(session->procFrame);
-#else
size_t fr = session->frameCount - session->framesIn;
if (inBuffer->frameCount < fr) {
fr = inBuffer->frameCount;
@@ -1696,7 +1164,6 @@
return status;
}
outBuffer->frameCount = inBuffer->frameCount;
-#endif
if (session->outBufSize < session->framesOut + session->frameCount) {
int16_t* buf;
@@ -1713,30 +1180,7 @@
session->outBuf = buf;
}
-#ifdef WEBRTC_LEGACY
- if (session->outResampler != NULL) {
- spx_uint32_t frIn = session->apmFrameCount;
- spx_uint32_t frOut = session->frameCount;
- if (session->inChannelCount == 1) {
- speex_resampler_process_int(
- session->outResampler, 0, session->procFrame->data_, &frIn,
- session->outBuf + session->framesOut * session->outChannelCount, &frOut);
- } else {
- speex_resampler_process_interleaved_int(
- session->outResampler, session->procFrame->data_, &frIn,
- session->outBuf + session->framesOut * session->outChannelCount, &frOut);
- }
- session->framesOut += frOut;
- } else {
- memcpy(session->outBuf + session->framesOut * session->outChannelCount,
- session->procFrame->data_,
- session->frameCount * session->outChannelCount * sizeof(int16_t));
- session->framesOut += session->frameCount;
- }
- size_t fr = session->framesOut;
-#else
fr = session->framesOut;
-#endif
if (framesRq - framesWr < fr) {
fr = framesRq - framesWr;
}
@@ -2129,63 +1573,6 @@
if ((session->revProcessedMsk & session->revEnabledMsk) == session->revEnabledMsk) {
effect->session->revProcessedMsk = 0;
-#ifdef WEBRTC_LEGACY
- if (session->revResampler != NULL) {
- size_t fr = session->frameCount - session->framesRev;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- if (session->revBufSize < session->framesRev + fr) {
- int16_t* buf;
- session->revBufSize = session->framesRev + fr;
- buf = (int16_t*)realloc(
- session->revBuf,
- session->revBufSize * session->inChannelCount * sizeof(int16_t));
- if (buf == NULL) {
- session->framesRev = 0;
- free(session->revBuf);
- session->revBuf = NULL;
- return -ENOMEM;
- }
- session->revBuf = buf;
- }
- memcpy(session->revBuf + session->framesRev * session->inChannelCount, inBuffer->s16,
- fr * session->inChannelCount * sizeof(int16_t));
-
- session->framesRev += fr;
- inBuffer->frameCount = fr;
- if (session->framesRev < session->frameCount) {
- return 0;
- }
- spx_uint32_t frIn = session->framesRev;
- spx_uint32_t frOut = session->apmFrameCount;
- if (session->inChannelCount == 1) {
- speex_resampler_process_int(session->revResampler, 0, session->revBuf, &frIn,
- session->revFrame->data_, &frOut);
- } else {
- speex_resampler_process_interleaved_int(session->revResampler, session->revBuf,
- &frIn, session->revFrame->data_, &frOut);
- }
- memmove(session->revBuf, session->revBuf + frIn * session->inChannelCount,
- (session->framesRev - frIn) * session->inChannelCount * sizeof(int16_t));
- session->framesRev -= frIn;
- } else {
- size_t fr = session->frameCount - session->framesRev;
- if (inBuffer->frameCount < fr) {
- fr = inBuffer->frameCount;
- }
- memcpy(session->revFrame->data_ + session->framesRev * session->inChannelCount,
- inBuffer->s16, fr * session->inChannelCount * sizeof(int16_t));
- session->framesRev += fr;
- inBuffer->frameCount = fr;
- if (session->framesRev < session->frameCount) {
- return 0;
- }
- session->framesRev = 0;
- }
- session->revFrame->samples_per_channel_ = session->apmFrameCount;
- effect->session->apm->AnalyzeReverseStream(session->revFrame);
-#else
size_t fr = session->frameCount - session->framesRev;
if (inBuffer->frameCount < fr) {
fr = inBuffer->frameCount;
@@ -2205,7 +1592,6 @@
ALOGE("Process Reverse Stream failed with error %d\n", status);
return status;
}
-#endif
return 0;
} else {
return -ENODATA;
diff --git a/media/libeffects/preprocessing/benchmarks/Android.bp b/media/libeffects/preprocessing/benchmarks/Android.bp
index 2808293..262fd19 100644
--- a/media/libeffects/preprocessing/benchmarks/Android.bp
+++ b/media/libeffects/preprocessing/benchmarks/Android.bp
@@ -1,31 +1,4 @@
cc_benchmark {
- name: "preprocessing_legacy_benchmark",
- vendor: true,
- relative_install_path: "soundfx",
- srcs: ["preprocessing_benchmark.cpp"],
- shared_libs: [
- "libaudiopreprocessing_legacy",
- "libaudioutils",
- "liblog",
- "libutils",
- "libwebrtc_audio_preprocessing",
- ],
- cflags: [
- "-DWEBRTC_POSIX",
- "-DWEBRTC_LEGACY",
- "-fvisibility=default",
- "-Wall",
- "-Werror",
- "-Wextra",
- ],
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- "libwebrtc_absl_headers",
- ],
-}
-
-cc_benchmark {
name: "preprocessing_benchmark",
vendor: true,
relative_install_path: "soundfx",
diff --git a/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp b/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp
index 3a0ad6d..694a6c4 100644
--- a/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp
+++ b/media/libeffects/preprocessing/benchmarks/preprocessing_benchmark.cpp
@@ -54,9 +54,7 @@
#include <cstdlib>
#include <random>
#include <vector>
-#ifndef WEBRTC_LEGACY
#include <audio_effects/effect_agc2.h>
-#endif
#include <audio_effects/effect_ns.h>
#include <benchmark/benchmark.h>
#include <hardware/audio_effect.h>
@@ -76,10 +74,8 @@
{0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
// ns uuid
{0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}},
-#ifndef WEBRTC_LEGACY
// agc2 uuid
{0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}},
-#endif
};
constexpr size_t kNumEffectUuids = std::size(kEffectUuids);
constexpr audio_channel_mask_t kChMasks[] = {
@@ -93,9 +89,7 @@
PREPROC_AGC, // Automatic Gain Control
PREPROC_AEC, // Acoustic Echo Canceler
PREPROC_NS, // Noise Suppressor
-#ifndef WEBRTC_LEGACY
PREPROC_AGC2, // Automatic Gain Control 2
-#endif
PREPROC_NUM_EFFECTS
};
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
index 045b0d3..b439880 100644
--- a/media/libeffects/preprocessing/tests/Android.bp
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -1,37 +1,5 @@
// audio preprocessing unit test
cc_test {
- name: "AudioPreProcessingLegacyTest",
-
- vendor: true,
-
- relative_install_path: "soundfx",
-
- srcs: ["PreProcessingTest.cpp"],
-
- shared_libs: [
- "libaudiopreprocessing_legacy",
- "libaudioutils",
- "liblog",
- "libutils",
- "libwebrtc_audio_preprocessing",
- ],
-
- cflags: [
- "-DWEBRTC_POSIX",
- "-DWEBRTC_LEGACY",
- "-fvisibility=default",
- "-Wall",
- "-Werror",
- "-Wextra",
- ],
-
- header_libs: [
- "libaudioeffects",
- "libhardware_headers",
- ],
-}
-
-cc_test {
name: "AudioPreProcessingTest",
vendor: true,
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
index 65b9469..5f223c9 100644
--- a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -22,9 +22,7 @@
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_agc.h>
-#ifndef WEBRTC_LEGACY
#include <audio_effects/effect_agc2.h>
-#endif
#include <audio_effects/effect_ns.h>
#include <log/log.h>
@@ -38,9 +36,7 @@
// types of pre processing modules
enum PreProcId {
PREPROC_AGC, // Automatic Gain Control
-#ifndef WEBRTC_LEGACY
PREPROC_AGC2, // Automatic Gain Control 2
-#endif
PREPROC_AEC, // Acoustic Echo Canceler
PREPROC_NS, // Noise Suppressor
PREPROC_NUM_EFFECTS
@@ -57,11 +53,9 @@
ARG_AGC_COMP_LVL,
ARG_AEC_DELAY,
ARG_NS_LVL,
-#ifndef WEBRTC_LEGACY
ARG_AGC2_GAIN,
ARG_AGC2_LVL,
ARG_AGC2_SAT_MGN
-#endif
};
struct preProcConfigParams_t {
@@ -70,19 +64,15 @@
int nsLevel = 0; // a value between 0-3
int agcTargetLevel = 3; // in dB
int agcCompLevel = 9; // in dB
-#ifndef WEBRTC_LEGACY
float agc2Gain = 0.f; // in dB
float agc2SaturationMargin = 2.f; // in dB
int agc2Level = 0; // either kRms(0) or kPeak(1)
-#endif
int aecDelay = 0; // in ms
};
const effect_uuid_t kPreProcUuids[PREPROC_NUM_EFFECTS] = {
{0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // agc uuid
-#ifndef WEBRTC_LEGACY
{0x89f38e65, 0xd4d2, 0x4d64, 0xad0e, {0x2b, 0x3e, 0x79, 0x9e, 0xa8, 0x86}}, // agc2 uuid
-#endif
{0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // aec uuid
{0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // ns uuid
};
@@ -138,24 +128,20 @@
printf("\n Enable Noise Suppression, default disabled");
printf("\n --agc");
printf("\n Enable Gain Control, default disabled");
-#ifndef WEBRTC_LEGACY
printf("\n --agc2");
printf("\n Enable Gain Controller 2, default disabled");
-#endif
printf("\n --ns_lvl <ns_level>");
printf("\n Noise Suppression level in dB, default value 0dB");
printf("\n --agc_tgt_lvl <target_level>");
printf("\n AGC Target Level in dB, default value 3dB");
printf("\n --agc_comp_lvl <comp_level>");
printf("\n AGC Comp Level in dB, default value 9dB");
-#ifndef WEBRTC_LEGACY
printf("\n --agc2_gain <fixed_digital_gain>");
printf("\n AGC Fixed Digital Gain in dB, default value 0dB");
printf("\n --agc2_lvl <level_estimator>");
printf("\n AGC Adaptive Digital Level Estimator, default value kRms");
printf("\n --agc2_sat_mgn <saturation_margin>");
printf("\n AGC Adaptive Digital Saturation Margin in dB, default value 2dB");
-#endif
printf("\n --aec_delay <delay>");
printf("\n AEC delay value in ms, default value 0ms");
printf("\n");
@@ -217,18 +203,14 @@
{"ch_mask", required_argument, nullptr, ARG_CH_MASK},
{"agc_tgt_lvl", required_argument, nullptr, ARG_AGC_TGT_LVL},
{"agc_comp_lvl", required_argument, nullptr, ARG_AGC_COMP_LVL},
-#ifndef WEBRTC_LEGACY
{"agc2_gain", required_argument, nullptr, ARG_AGC2_GAIN},
{"agc2_lvl", required_argument, nullptr, ARG_AGC2_LVL},
{"agc2_sat_mgn", required_argument, nullptr, ARG_AGC2_SAT_MGN},
-#endif
{"aec_delay", required_argument, nullptr, ARG_AEC_DELAY},
{"ns_lvl", required_argument, nullptr, ARG_NS_LVL},
{"aec", no_argument, &effectEn[PREPROC_AEC], 1},
{"agc", no_argument, &effectEn[PREPROC_AGC], 1},
-#ifndef WEBRTC_LEGACY
{"agc2", no_argument, &effectEn[PREPROC_AGC2], 1},
-#endif
{"ns", no_argument, &effectEn[PREPROC_NS], 1},
{nullptr, 0, nullptr, 0},
};
@@ -277,7 +259,6 @@
preProcCfgParams.agcCompLevel = atoi(optarg);
break;
}
-#ifndef WEBRTC_LEGACY
case ARG_AGC2_GAIN: {
preProcCfgParams.agc2Gain = atof(optarg);
break;
@@ -290,7 +271,6 @@
preProcCfgParams.agc2SaturationMargin = atof(optarg);
break;
}
-#endif
case ARG_AEC_DELAY: {
preProcCfgParams.aecDelay = atoi(optarg);
break;
@@ -387,7 +367,6 @@
return EXIT_FAILURE;
}
}
-#ifndef WEBRTC_LEGACY
if (effectEn[PREPROC_AGC2]) {
if (int status = preProcSetConfigParam(AGC2_PARAM_FIXED_DIGITAL_GAIN,
(float)preProcCfgParams.agc2Gain,
@@ -411,7 +390,6 @@
return EXIT_FAILURE;
}
}
-#endif
if (effectEn[PREPROC_NS]) {
if (int status = preProcSetConfigParam(NS_PARAM_LEVEL, (uint32_t)preProcCfgParams.nsLevel,
effectHandle[PREPROC_NS]);
diff --git a/media/libmedia/IMediaExtractor.cpp b/media/libmedia/IMediaExtractor.cpp
index 39caf53..7ed76d8 100644
--- a/media/libmedia/IMediaExtractor.cpp
+++ b/media/libmedia/IMediaExtractor.cpp
@@ -38,7 +38,8 @@
FLAGS,
SETMEDIACAS,
NAME,
- GETMETRICS
+ GETMETRICS,
+ SETENTRYPOINT
};
class BpMediaExtractor : public BpInterface<IMediaExtractor> {
@@ -142,6 +143,13 @@
}
return nm;
}
+
+ virtual status_t setEntryPoint(EntryPoint entryPoint) {
+ Parcel data, reply;
+ data.writeInterfaceToken(BpMediaExtractor::getInterfaceDescriptor());
+ data.writeInt32(static_cast<int32_t>(entryPoint));
+ return remote()->transact(SETENTRYPOINT, data, &reply);
+ }
};
IMPLEMENT_META_INTERFACE(MediaExtractor, "android.media.IMediaExtractor");
@@ -232,6 +240,16 @@
reply->writeString8(nm);
return NO_ERROR;
}
+ case SETENTRYPOINT: {
+ ALOGV("setEntryPoint");
+ CHECK_INTERFACE(IMediaExtractor, data, reply);
+ int32_t entryPoint;
+ status_t err = data.readInt32(&entryPoint);
+ if (err == OK) {
+ setEntryPoint(EntryPoint(entryPoint));
+ }
+ return err;
+ }
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmedia/include/android/IMediaExtractor.h b/media/libmedia/include/android/IMediaExtractor.h
index 3e035ad..f9cafde 100644
--- a/media/libmedia/include/android/IMediaExtractor.h
+++ b/media/libmedia/include/android/IMediaExtractor.h
@@ -63,6 +63,15 @@
virtual status_t setMediaCas(const HInterfaceToken &casToken) = 0;
virtual String8 name() = 0;
+
+ enum class EntryPoint {
+ SDK = 1,
+ NDK_WITH_JVM = 2,
+ NDK_NO_JVM = 3,
+ OTHER = 4,
+ };
+
+ virtual status_t setEntryPoint(EntryPoint entryPoint) = 0;
};
diff --git a/media/libmediatranscoding/transcoder/VideoTrackTranscoder.cpp b/media/libmediatranscoding/transcoder/VideoTrackTranscoder.cpp
index 5ec5e08..b4cc65f 100644
--- a/media/libmediatranscoding/transcoder/VideoTrackTranscoder.cpp
+++ b/media/libmediatranscoding/transcoder/VideoTrackTranscoder.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "VideoTrackTranscoder"
#include <android-base/logging.h>
+#include <android-base/properties.h>
#include <media/NdkCommon.h>
#include <media/VideoTrackTranscoder.h>
#include <utils/AndroidThreads.h>
@@ -39,7 +40,8 @@
// Default key frame interval in seconds.
static constexpr float kDefaultKeyFrameIntervalSeconds = 1.0f;
// Default codec operating rate.
-static constexpr int32_t kDefaultCodecOperatingRate = 240;
+static int32_t kDefaultCodecOperatingRate =
+ base::GetIntProperty("debug.media.transcoding.codec_max_operating_rate", /*default*/ 240);
// Default codec priority.
static constexpr int32_t kDefaultCodecPriority = 1;
// Default bitrate, in case source estimation fails.
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 0af97df..447d599 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -1137,11 +1137,24 @@
if (!truncatePreAllocation()) {
if (err == OK) { err = ERROR_IO; }
}
+
+ // TODO(b/174770856) remove this measurement (and perhaps the fsync)
+ nsecs_t sync_started = systemTime(SYSTEM_TIME_REALTIME);
if (fsync(mFd) != 0) {
ALOGW("(ignored)fsync err:%s(%d)", std::strerror(errno), errno);
// Don't bubble up fsync error, b/157291505.
// if (err == OK) { err = ERROR_IO; }
}
+ nsecs_t sync_finished = systemTime(SYSTEM_TIME_REALTIME);
+ nsecs_t sync_elapsed_ns = sync_finished - sync_started;
+ int64_t filesize = -1;
+ struct stat statbuf;
+ if (fstat(mFd, &statbuf) == 0) {
+ filesize = statbuf.st_size;
+ }
+ ALOGD("final fsync() takes %" PRId64 " ms, file size %" PRId64,
+ sync_elapsed_ns / 1000000, (int64_t) filesize);
+
if (close(mFd) != 0) {
ALOGE("close err:%s(%d)", std::strerror(errno), errno);
if (err == OK) { err = ERROR_IO; }
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 6245014..f2c7dd6 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -50,8 +50,9 @@
mSampleTimeUs(timeUs) {
}
-NuMediaExtractor::NuMediaExtractor()
- : mTotalBitrate(-1LL),
+NuMediaExtractor::NuMediaExtractor(EntryPoint entryPoint)
+ : mEntryPoint(entryPoint),
+ mTotalBitrate(-1LL),
mDurationUs(-1LL) {
}
@@ -93,6 +94,7 @@
if (mImpl == NULL) {
return ERROR_UNSUPPORTED;
}
+ setEntryPointToRemoteMediaExtractor();
status_t err = OK;
if (!mCasToken.empty()) {
@@ -134,6 +136,7 @@
if (mImpl == NULL) {
return ERROR_UNSUPPORTED;
}
+ setEntryPointToRemoteMediaExtractor();
if (!mCasToken.empty()) {
err = mImpl->setMediaCas(mCasToken);
@@ -168,6 +171,7 @@
if (mImpl == NULL) {
return ERROR_UNSUPPORTED;
}
+ setEntryPointToRemoteMediaExtractor();
if (!mCasToken.empty()) {
err = mImpl->setMediaCas(mCasToken);
@@ -489,6 +493,16 @@
}
}
+void NuMediaExtractor::setEntryPointToRemoteMediaExtractor() {
+ if (mImpl == NULL) {
+ return;
+ }
+ status_t err = mImpl->setEntryPoint(mEntryPoint);
+ if (err != OK) {
+ ALOGW("Failed to set entry point with error %d.", err);
+ }
+}
+
ssize_t NuMediaExtractor::fetchAllTrackSamples(
int64_t seekTimeUs, MediaSource::ReadOptions::SeekMode mode) {
TrackInfo *minInfo = NULL;
diff --git a/media/libstagefright/RemoteMediaExtractor.cpp b/media/libstagefright/RemoteMediaExtractor.cpp
index 25e43c2..381eb1a 100644
--- a/media/libstagefright/RemoteMediaExtractor.cpp
+++ b/media/libstagefright/RemoteMediaExtractor.cpp
@@ -39,6 +39,12 @@
static const char *kExtractorFormat = "android.media.mediaextractor.fmt";
static const char *kExtractorMime = "android.media.mediaextractor.mime";
static const char *kExtractorTracks = "android.media.mediaextractor.ntrk";
+static const char *kExtractorEntryPoint = "android.media.mediaextractor.entry";
+
+static const char *kEntryPointSdk = "sdk";
+static const char *kEntryPointWithJvm = "ndk-with-jvm";
+static const char *kEntryPointNoJvm = "ndk-no-jvm";
+static const char *kEntryPointOther = "other";
RemoteMediaExtractor::RemoteMediaExtractor(
MediaExtractor *extractor,
@@ -74,6 +80,9 @@
}
// what else is interesting and not already available?
}
+ // By default, we set the entry point to be "other". Clients of this
+ // class will override this value by calling setEntryPoint.
+ mMetricsItem->setCString(kExtractorEntryPoint, kEntryPointOther);
}
}
@@ -143,6 +152,28 @@
return String8(mExtractor->name());
}
+status_t RemoteMediaExtractor::setEntryPoint(EntryPoint entryPoint) {
+ const char* entryPointString;
+ switch (entryPoint) {
+ case EntryPoint::SDK:
+ entryPointString = kEntryPointSdk;
+ break;
+ case EntryPoint::NDK_WITH_JVM:
+ entryPointString = kEntryPointWithJvm;
+ break;
+ case EntryPoint::NDK_NO_JVM:
+ entryPointString = kEntryPointNoJvm;
+ break;
+ case EntryPoint::OTHER:
+ entryPointString = kEntryPointOther;
+ break;
+ default:
+ return BAD_VALUE;
+ }
+ mMetricsItem->setCString(kExtractorEntryPoint, entryPointString);
+ return OK;
+}
+
////////////////////////////////////////////////////////////////////////////////
// static
diff --git a/media/libstagefright/bqhelper/Android.bp b/media/libstagefright/bqhelper/Android.bp
index 2b0494c..8698d33 100644
--- a/media/libstagefright/bqhelper/Android.bp
+++ b/media/libstagefright/bqhelper/Android.bp
@@ -101,7 +101,6 @@
"//apex_available:platform",
],
vendor_available: false,
- min_sdk_version: "29",
static_libs: [
"libgui_bufferqueue_static",
],
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index f242b19..7752bda 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -33,7 +33,7 @@
#include <media/stagefright/foundation/hexdump.h>
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
#include <binder/Parcel.h>
#endif
@@ -646,7 +646,7 @@
return s;
}
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
// static
sp<AMessage> AMessage::FromParcel(const Parcel &parcel, size_t maxNestingLevel) {
int32_t what = parcel.readInt32();
@@ -813,7 +813,7 @@
}
}
}
-#endif // !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#endif // __ANDROID_VNDK__
sp<AMessage> AMessage::changesFrom(const sp<const AMessage> &other, bool deep) const {
if (other == NULL) {
diff --git a/media/libstagefright/foundation/AString.cpp b/media/libstagefright/foundation/AString.cpp
index b1ed077..8722e14 100644
--- a/media/libstagefright/foundation/AString.cpp
+++ b/media/libstagefright/foundation/AString.cpp
@@ -27,7 +27,7 @@
#include "ADebug.h"
#include "AString.h"
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
#include <binder/Parcel.h>
#endif
@@ -365,7 +365,7 @@
return !strcasecmp(mData + mSize - suffixLen, suffix);
}
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
// static
AString AString::FromParcel(const Parcel &parcel) {
size_t size = static_cast<size_t>(parcel.readInt32());
@@ -380,7 +380,7 @@
}
return err;
}
-#endif // !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#endif
AString AStringPrintf(const char *format, ...) {
va_list ap;
diff --git a/media/libstagefright/foundation/Android.bp b/media/libstagefright/foundation/Android.bp
index 39670a2..ebf1035 100644
--- a/media/libstagefright/foundation/Android.bp
+++ b/media/libstagefright/foundation/Android.bp
@@ -86,11 +86,6 @@
"-DNO_IMEMORY",
],
},
- apex: {
- exclude_shared_libs: [
- "libbinder",
- ],
- },
darwin: {
enabled: false,
},
diff --git a/media/libstagefright/foundation/MediaBuffer.cpp b/media/libstagefright/foundation/MediaBuffer.cpp
index 68df21f..8e245dc 100644
--- a/media/libstagefright/foundation/MediaBuffer.cpp
+++ b/media/libstagefright/foundation/MediaBuffer.cpp
@@ -51,12 +51,12 @@
mRangeLength(size),
mOwnsData(true),
mMetaData(new MetaDataBase) {
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
if (size < kSharedMemThreshold
|| std::atomic_load_explicit(&mUseSharedMemory, std::memory_order_seq_cst) == 0) {
#endif
mData = malloc(size);
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
} else {
ALOGV("creating memoryDealer");
size_t newSize = 0;
diff --git a/media/libstagefright/foundation/MediaBufferGroup.cpp b/media/libstagefright/foundation/MediaBufferGroup.cpp
index fc98f28..3c25047 100644
--- a/media/libstagefright/foundation/MediaBufferGroup.cpp
+++ b/media/libstagefright/foundation/MediaBufferGroup.cpp
@@ -62,7 +62,7 @@
mInternal->mGrowthLimit = buffers;
}
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
if (buffer_size >= kSharedMemoryThreshold) {
ALOGD("creating MemoryDealer");
// Using a single MemoryDealer is efficient for a group of shared memory objects.
diff --git a/media/libstagefright/foundation/MetaData.cpp b/media/libstagefright/foundation/MetaData.cpp
index 7f48cfd..8174597 100644
--- a/media/libstagefright/foundation/MetaData.cpp
+++ b/media/libstagefright/foundation/MetaData.cpp
@@ -28,7 +28,7 @@
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/MetaData.h>
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
#include <binder/Parcel.h>
#endif
@@ -48,7 +48,7 @@
MetaData::~MetaData() {
}
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
/* static */
sp<MetaData> MetaData::createFromParcel(const Parcel &parcel) {
diff --git a/media/libstagefright/foundation/MetaDataBase.cpp b/media/libstagefright/foundation/MetaDataBase.cpp
index 3f050ea..4b439c6 100644
--- a/media/libstagefright/foundation/MetaDataBase.cpp
+++ b/media/libstagefright/foundation/MetaDataBase.cpp
@@ -28,7 +28,7 @@
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/MetaDataBase.h>
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
#include <binder/Parcel.h>
#endif
@@ -452,7 +452,7 @@
}
}
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
status_t MetaDataBase::writeToParcel(Parcel &parcel) {
status_t ret;
size_t numItems = mInternalData->mItems.size();
@@ -532,7 +532,7 @@
ALOGW("no metadata in parcel");
return UNKNOWN_ERROR;
}
-#endif // !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#endif
} // namespace android
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/AMessage.h b/media/libstagefright/foundation/include/media/stagefright/foundation/AMessage.h
index 31e58ba..b5d6666 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/AMessage.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/AMessage.h
@@ -63,7 +63,7 @@
AMessage();
AMessage(uint32_t what, const sp<const AHandler> &handler);
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
// Construct an AMessage from a parcel.
// nestingAllowed determines how many levels AMessage can be nested inside
// AMessage. The default value here is arbitrarily set to 255.
@@ -88,7 +88,7 @@
// All items in the AMessage must have types that are recognized by
// FromParcel(); otherwise, TRESPASS error will occur.
void writeToParcel(Parcel *parcel) const;
-#endif // !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#endif
void setWhat(uint32_t what);
uint32_t what() const;
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/AString.h b/media/libstagefright/foundation/include/media/stagefright/foundation/AString.h
index 517774b..deef0d4 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/AString.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/AString.h
@@ -89,7 +89,7 @@
void tolower();
-#if !defined(__ANDROID_VNDK__) && !defined(__ANDROID_APEX__)
+#ifndef __ANDROID_VNDK__
static AString FromParcel(const Parcel &parcel);
status_t writeToParcel(Parcel *parcel) const;
#endif
diff --git a/media/libstagefright/include/media/stagefright/MediaBuffer.h b/media/libstagefright/include/media/stagefright/MediaBuffer.h
index 2c03f27..9145b63 100644
--- a/media/libstagefright/include/media/stagefright/MediaBuffer.h
+++ b/media/libstagefright/include/media/stagefright/MediaBuffer.h
@@ -46,7 +46,7 @@
explicit MediaBuffer(size_t size);
explicit MediaBuffer(const sp<ABuffer> &buffer);
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
MediaBuffer(const sp<IMemory> &mem) :
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
@@ -97,7 +97,7 @@
}
virtual int remoteRefcount() const {
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
@@ -114,7 +114,7 @@
// returns old value
int addRemoteRefcount(int32_t value) {
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
@@ -132,7 +132,7 @@
}
static bool isDeadObject(const sp<IMemory> &memory) {
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
@@ -235,7 +235,7 @@
};
inline SharedControl *getSharedControl() const {
-#if !defined(NO_IMEMORY) && !defined(__ANDROID_APEX__)
+#ifndef NO_IMEMORY
// TODO: Using unsecurePointer() has some associated security pitfalls
// (see declaration for details).
// Either document why it is safe in this case or address the
diff --git a/media/libstagefright/include/media/stagefright/NuMediaExtractor.h b/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
index 227cead..d8f2b00 100644
--- a/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
+++ b/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
@@ -47,12 +47,14 @@
SAMPLE_FLAG_ENCRYPTED = 2,
};
+ typedef IMediaExtractor::EntryPoint EntryPoint;
+
// identical to IMediaExtractor::GetTrackMetaDataFlags
enum GetTrackFormatFlags {
kIncludeExtensiveMetaData = 1, // reads sample table and possibly stream headers
};
- NuMediaExtractor();
+ explicit NuMediaExtractor(EntryPoint entryPoint);
status_t setDataSource(
const sp<MediaHTTPService> &httpService,
@@ -128,6 +130,8 @@
uint32_t mTrackFlags; // bitmask of "TrackFlags"
};
+ const EntryPoint mEntryPoint;
+
mutable Mutex mLock;
sp<DataSource> mDataSource;
@@ -139,6 +143,8 @@
int64_t mTotalBitrate; // in bits/sec
int64_t mDurationUs;
+ void setEntryPointToRemoteMediaExtractor();
+
ssize_t fetchAllTrackSamples(
int64_t seekTimeUs = -1ll,
MediaSource::ReadOptions::SeekMode mode =
diff --git a/media/libstagefright/include/media/stagefright/RemoteMediaExtractor.h b/media/libstagefright/include/media/stagefright/RemoteMediaExtractor.h
index 2ce7bc7..25125f2 100644
--- a/media/libstagefright/include/media/stagefright/RemoteMediaExtractor.h
+++ b/media/libstagefright/include/media/stagefright/RemoteMediaExtractor.h
@@ -42,6 +42,7 @@
virtual uint32_t flags() const;
virtual status_t setMediaCas(const HInterfaceToken &casToken);
virtual String8 name();
+ virtual status_t setEntryPoint(EntryPoint entryPoint);
private:
MediaExtractor *mExtractor;
diff --git a/media/libstagefright/renderfright/include/renderengine/RenderEngine.h b/media/libstagefright/renderfright/include/renderengine/RenderEngine.h
index 09a0f65..40fdff4 100644
--- a/media/libstagefright/renderfright/include/renderengine/RenderEngine.h
+++ b/media/libstagefright/renderfright/include/renderengine/RenderEngine.h
@@ -33,7 +33,7 @@
/**
* Allows to set RenderEngine backend to GLES (default) or Vulkan (NOT yet supported).
*/
-#define PROPERTY_DEBUG_RENDERENGINE_BACKEND "debug.renderengine.backend"
+#define PROPERTY_DEBUG_RENDERENGINE_BACKEND "debug.stagefright.renderengine.backend"
struct ANativeWindowBuffer;
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index f57077c..07f9dd3 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -131,7 +131,7 @@
unsigned start = (unsigned)((rand()* 1000LL)/RAND_MAX) + 15550;
start &= ~1;
- for (unsigned port = start; port < 65536; port += 2) {
+ for (unsigned port = start; port < 65535; port += 2) {
struct sockaddr_in addr;
memset(addr.sin_zero, 0, sizeof(addr.sin_zero));
addr.sin_family = AF_INET;
@@ -149,6 +149,13 @@
(const struct sockaddr *)&addr, sizeof(addr)) == 0) {
*rtpPort = port;
return;
+ } else {
+ // we should recreate a RTP socket to avoid bind other port in same RTP socket
+ close(*rtpSocket);
+
+ *rtpSocket = socket(AF_INET, SOCK_DGRAM, 0);
+ CHECK_GE(*rtpSocket, 0);
+ bumpSocketBufferSize(*rtpSocket);
}
}
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index 0da0740..0c65e9e 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -22,6 +22,7 @@
#include <media/NdkMediaExtractor.h>
#include <media/NdkMediaErrorPriv.h>
#include <media/NdkMediaFormatPriv.h>
+#include "NdkJavaVMHelperPriv.h"
#include "NdkMediaDataSourcePriv.h"
@@ -63,7 +64,10 @@
AMediaExtractor* AMediaExtractor_new() {
ALOGV("ctor");
AMediaExtractor *mData = new AMediaExtractor();
- mData->mImpl = new NuMediaExtractor();
+ mData->mImpl = new NuMediaExtractor(
+ NdkJavaVMHelper::getJNIEnv() != nullptr
+ ? NuMediaExtractor::EntryPoint::NDK_WITH_JVM
+ : NuMediaExtractor::EntryPoint::NDK_NO_JVM );
return mData;
}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d442af6..78ad467 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2593,7 +2593,7 @@
sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
- aidl2legacy_audio_output_flags_mask(request.flags));
+ aidl2legacy_int32_t_audio_output_flags_t_mask(request.flags));
audio_io_handle_t output;
uint32_t latencyMs;
@@ -2643,7 +2643,8 @@
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
- response->flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_mask(flags));
+ response->flags = VALUE_OR_RETURN_STATUS(
+ legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
return NO_ERROR;
}
@@ -2819,7 +2820,7 @@
device.mType,
device.address().c_str(),
VALUE_OR_RETURN_STATUS(aidl2legacy_AudioSourceType_audio_source_t(request.source)),
- VALUE_OR_RETURN_STATUS(aidl2legacy_audio_input_flags_mask(request.flags)),
+ VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_input_flags_t_mask(request.flags)),
AUDIO_DEVICE_NONE,
String8{});
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index eaad6ef..3ab7737 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -60,6 +60,7 @@
namespace android {
+using aidl_utils::statusTFromBinderStatus;
using binder::Status;
namespace {
@@ -3027,7 +3028,7 @@
bs = handle.second->disable(&status);
}
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
}
}
}
@@ -3142,7 +3143,7 @@
bs = (*handle)->disable(&status);
}
if (!bs.isOk()) {
- status = bs.transactionError();
+ status = statusTFromBinderStatus(bs);
}
}
return status;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 68b709f..6049f62 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -54,6 +54,7 @@
namespace android {
+using aidl_utils::binderStatusFromStatusT;
using binder::Status;
using media::VolumeShaper;
// ----------------------------------------------------------------------------
@@ -371,7 +372,7 @@
if (*_aidl_return != OK) {
return Status::ok();
}
- *timestamp = legacy2aidl_AudioTimestamp(legacy).value();
+ *timestamp = legacy2aidl_AudioTimestamp_AudioTimestampInternal(legacy).value();
return Status::ok();
}
@@ -2132,7 +2133,7 @@
binder::Status AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
int /*audio_session_t*/ triggerSession) {
ALOGV("%s()", __func__);
- return binder::Status::fromStatusT(
+ return binderStatusFromStatusT(
mRecordTrack->start((AudioSystem::sync_event_t)event, (audio_session_t) triggerSession));
}
@@ -2155,19 +2156,19 @@
for (size_t i = 0; status == OK && i < mics.size(); ++i) {
status = mics[i].writeToParcelable(&activeMicrophones->at(i));
}
- return binder::Status::fromStatusT(status);
+ return binderStatusFromStatusT(status);
}
binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneDirection(
int /*audio_microphone_direction_t*/ direction) {
ALOGV("%s()", __func__);
- return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneDirection(
+ return binderStatusFromStatusT(mRecordTrack->setPreferredMicrophoneDirection(
static_cast<audio_microphone_direction_t>(direction)));
}
binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneFieldDimension(float zoom) {
ALOGV("%s()", __func__);
- return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneFieldDimension(zoom));
+ return binderStatusFromStatusT(mRecordTrack->setPreferredMicrophoneFieldDimension(zoom));
}
// ----------------------------------------------------------------------------
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index dd128a2..90b93e2 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -57,7 +57,7 @@
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
- request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_mask(flags));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
status_t status = af->openOutput(request, &response);
if (status == OK) {
@@ -134,7 +134,7 @@
request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
- request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_mask(flags));
+ request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
media::OpenInputResponse response;
status_t status = af->openInput(request, &response);
diff --git a/services/mediametrics/statsd_extractor.cpp b/services/mediametrics/statsd_extractor.cpp
index 16814d9..4180e0c 100644
--- a/services/mediametrics/statsd_extractor.cpp
+++ b/services/mediametrics/statsd_extractor.cpp
@@ -71,6 +71,22 @@
metrics_proto.set_tracks(ntrk);
}
+ // android.media.mediaextractor.entry string
+ std::string entry_point_string;
+ if (item->getString("android.media.mediaextractor.entry", &entry_point_string)) {
+ stats::mediametrics::ExtractorData::EntryPoint entry_point;
+ if (entry_point_string == "sdk") {
+ entry_point = stats::mediametrics::ExtractorData_EntryPoint_SDK;
+ } else if (entry_point_string == "ndk-with-jvm") {
+ entry_point = stats::mediametrics::ExtractorData_EntryPoint_NDK_WITH_JVM;
+ } else if (entry_point_string == "ndk-no-jvm") {
+ entry_point = stats::mediametrics::ExtractorData_EntryPoint_NDK_NO_JVM;
+ } else {
+ entry_point = stats::mediametrics::ExtractorData_EntryPoint_OTHER;
+ }
+ metrics_proto.set_entry_point(entry_point);
+ }
+
std::string serialized;
if (!metrics_proto.SerializeToString(&serialized)) {
ALOGE("Failed to serialize extractor metrics");