Merge "lvm wrapper: Set SourceFormat as per number of channels"
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/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/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/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/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/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");