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Bug 1059573 - Add an option to use hardware AEC for WebRTC. r=jesup,ted

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This commit is contained in:
Gian-Carlo Pascutto 2014-09-24 19:11:58 +02:00
parent d45d23a70b
commit 2d486eeb3a
8 changed files with 184 additions and 6 deletions
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4
build/gyp.mozbuild
View File

@ -45,9 +45,13 @@ gyp_vars = {
# (for vp8) chromium sets to 0 also
'use_temporal_layers': 0,
# Creates AEC internal sample dump files in current directory
'aec_debug_dump': 1,
# Enable and force use of hardware AEC
'hardware_aec_ns': 1 if CONFIG['MOZ_WEBRTC_HARDWARE_AEC_NS'] else 0,
# codec enable/disables:
'include_g711': 1,
'include_opus': 1,

4
config/system-headers
View File

@ -200,6 +200,8 @@ atlwin.cpp
ATSTypes.h
ATSUnicode.h
#ifdef ANDROID
audio_effects/effect_aec.h
audio_effects/effect_ns.h
AudioParameter.h
AudioSystem.h
AudioTrack.h
@ -729,6 +731,8 @@ Math64.h
math.h
mbstring.h
#ifdef ANDROID
media/AudioEffect.h
media/AudioSystem.h
media/ICrypto.h
media/IOMX.h
media/MediaProfiles.h

14
configure.in
View File

@ -3793,6 +3793,7 @@ MOZ_PEERCONNECTION=
MOZ_SRTP=
MOZ_WEBRTC_SIGNALING=
MOZ_WEBRTC_ASSERT_ALWAYS=1
MOZ_WEBRTC_HARDWARE_AEC_NS=
MOZ_SCTP=
MOZ_ANDROID_OMX=
MOZ_MEDIA_NAVIGATOR=
@ -5006,11 +5007,24 @@ else
MOZ_SYNTH_PICO=
fi
dnl ========================================================
dnl = Force hardware AEC, disable webrtc.org AEC
dnl ========================================================
MOZ_ARG_ENABLE_BOOL(hardware-aec-ns,
[ --enable-hardware-aec-ns Enable support for hardware AEC and noise suppression],
MOZ_WEBRTC_HARDWARE_AEC_NS=1,
MOZ_WEBRTC_HARDWARE_AEC_NS=)
if test -n "$MOZ_WEBRTC_HARDWARE_AEC_NS"; then
AC_DEFINE(MOZ_WEBRTC_HARDWARE_AEC_NS)
fi
AC_SUBST(MOZ_WEBRTC)
AC_SUBST(MOZ_WEBRTC_LEAKING_TESTS)
AC_SUBST(MOZ_WEBRTC_SIGNALING)
AC_SUBST(MOZ_PEERCONNECTION)
AC_SUBST(MOZ_WEBRTC_ASSERT_ALWAYS)
AC_SUBST(MOZ_WEBRTC_HARDWARE_AEC_NS)
AC_SUBST(MOZ_SCTP)
AC_SUBST(MOZ_SRTP)
AC_SUBST_LIST(MOZ_WEBRTC_X11_LIBS)

14
media/libcubeb/src/android/sles_definitions.h
View File

@ -33,14 +33,20 @@
/** Audio recording preset values */
/** preset "none" cannot be set, it is used to indicate the current settings
* do not match any of the presets. */
#define SL_ANDROID_RECORDING_PRESET_NONE ((SLuint32) 0x00000000)
#define SL_ANDROID_RECORDING_PRESET_NONE ((SLuint32) 0x00000000)
/** generic recording configuration on the platform */
#define SL_ANDROID_RECORDING_PRESET_GENERIC ((SLuint32) 0x00000001)
#define SL_ANDROID_RECORDING_PRESET_GENERIC ((SLuint32) 0x00000001)
/** uses the microphone audio source with the same orientation as the camera
* if available, the main device microphone otherwise */
#define SL_ANDROID_RECORDING_PRESET_CAMCORDER ((SLuint32) 0x00000002)
#define SL_ANDROID_RECORDING_PRESET_CAMCORDER ((SLuint32) 0x00000002)
/** uses the main microphone tuned for voice recognition */
#define SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION ((SLuint32) 0x00000003)
#define SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION ((SLuint32) 0x00000003)
/** uses the main microphone tuned for audio communications */
#define SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION ((SLuint32) 0x00000004)
/** Audio recording get session ID (read only) */
/** Audio recording get session ID key */
#define SL_ANDROID_KEY_RECORDING_SESSION_ID ((const SLchar*) "androidRecordingSessionId")
/*---------------------------------------------------------------------------*/
/* Android AudioPlayer configuration */

126
media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.cc
View File

@ -22,6 +22,13 @@
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
#include <media/AudioSystem.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include <utils/Errors.h>
#endif
#define VOID_RETURN
#define OPENSL_RETURN_ON_FAILURE(op, ret_val) \
do { \
@ -65,6 +72,10 @@ OpenSlesInput::OpenSlesInput(
active_queue_(0),
rec_sampling_rate_(0),
agc_enabled_(false),
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
aec_(NULL),
ns_(NULL),
#endif
recording_delay_(0),
opensles_lib_(NULL) {
}
@ -390,6 +401,104 @@ bool OpenSlesInput::EnqueueAllBuffers() {
return true;
}
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
bool OpenSlesInput::CheckPlatformAEC() {
effect_descriptor_t fxDesc;
uint32_t numFx;
if (android::AudioEffect::queryNumberEffects(&numFx) != android::NO_ERROR) {
return false;
}
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "Platform has %d effects", numFx);
for (uint32_t i = 0; i < numFx; i++) {
if (android::AudioEffect::queryEffect(i, &fxDesc) != android::NO_ERROR) {
continue;
}
if (memcmp(&fxDesc.type, FX_IID_AEC, sizeof(fxDesc.type)) == 0) {
return true;
}
}
return false;
}
void OpenSlesInput::SetupVoiceMode() {
SLAndroidConfigurationItf configItf;
SLresult res = (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_ANDROIDCONFIGURATION_,
(void*)&configItf);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL GetInterface: %d", res);
if (res == SL_RESULT_SUCCESS) {
SLuint32 voiceMode = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
SLuint32 voiceSize = sizeof(voiceMode);
res = (*configItf)->SetConfiguration(configItf,
SL_ANDROID_KEY_RECORDING_PRESET,
&voiceMode, voiceSize);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL Set Voice mode res: %d", res);
}
}
void OpenSlesInput::SetupAECAndNS() {
bool hasAec = CheckPlatformAEC();
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "Platform has AEC: %d", hasAec);
// This code should not have been enabled if this fails, because it means the
// software AEC has will have been disabled as well. If you hit this, you need
// to fix your B2G config or fix the hardware AEC on your device.
assert(hasAec);
SLAndroidConfigurationItf configItf;
SLresult res = (*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_ANDROIDCONFIGURATION_,
(void*)&configItf);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL GetInterface: %d", res);
if (res == SL_RESULT_SUCCESS) {
SLuint32 sessionId = 0;
SLuint32 idSize = sizeof(sessionId);
res = (*configItf)->GetConfiguration(configItf,
SL_ANDROID_KEY_RECORDING_SESSION_ID,
&idSize, &sessionId);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL Get sessionId res: %d", res);
if (res == SL_RESULT_SUCCESS && idSize == sizeof(sessionId)) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL sessionId: %d", sessionId);
aec_ = new android::AudioEffect(FX_IID_AEC, NULL, 0, 0, 0, sessionId, 0);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL aec: %p", aec_);
if (aec_) {
android::status_t status = aec_->initCheck();
if (status == android::NO_ERROR || status == android::ALREADY_EXISTS) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL aec enabled");
aec_->setEnabled(true);
} else {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL aec disabled: %d", status);
delete aec_;
aec_ = NULL;
}
}
ns_ = new android::AudioEffect(FX_IID_NS, NULL, 0, 0, 0, sessionId, 0);
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL ns: %p", ns_);
if (ns_) {
android::status_t status = ns_->initCheck();
if (status == android::NO_ERROR || status == android::ALREADY_EXISTS) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL ns enabled");
ns_->setEnabled(true);
} else {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, id_, "OpenSL ns disabled: %d", status);
delete ns_;
ns_ = NULL;
}
}
}
}
}
#endif
bool OpenSlesInput::CreateAudioRecorder() {
if (!event_.Start()) {
assert(false);
@ -425,10 +534,19 @@ bool OpenSlesInput::CreateAudioRecorder() {
req),
false);
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
SetupVoiceMode();
#endif
// Realize the recorder in synchronous mode.
OPENSL_RETURN_ON_FAILURE((*sles_recorder_)->Realize(sles_recorder_,
SL_BOOLEAN_FALSE),
false);
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
SetupAECAndNS();
#endif
OPENSL_RETURN_ON_FAILURE(
(*sles_recorder_)->GetInterface(sles_recorder_, SL_IID_RECORD_,
static_cast<void*>(&sles_recorder_itf_)),
@ -444,6 +562,14 @@ bool OpenSlesInput::CreateAudioRecorder() {
void OpenSlesInput::DestroyAudioRecorder() {
event_.Stop();
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
delete aec_;
delete ns_;
aec_ = NULL;
ns_ = NULL;
#endif
if (sles_recorder_sbq_itf_) {
// Release all buffers currently queued up.
OPENSL_RETURN_ON_FAILURE(

11
media/webrtc/trunk/webrtc/modules/audio_device/android/opensles_input.h
View File

@ -17,6 +17,8 @@
#if !defined(WEBRTC_GONK)
#include "webrtc/modules/audio_device/android/audio_manager_jni.h"
#else
#include "media/AudioEffect.h"
#endif
#include "webrtc/modules/audio_device/android/low_latency_event.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
@ -143,6 +145,11 @@ class OpenSlesInput {
// etc, so it should be called when starting recording.
bool CreateAudioRecorder();
void DestroyAudioRecorder();
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
void SetupAECAndNS();
void SetupVoiceMode();
bool CheckPlatformAEC();
#endif
// When overrun happens there will be more frames received from OpenSL than
// the desired number of buffers. It is possible to expand the number of
@ -220,6 +227,10 @@ class OpenSlesInput {
uint32_t rec_sampling_rate_;
bool agc_enabled_;
#if defined(WEBRTC_GONK) && defined(WEBRTC_HARDWARE_AEC_NS)
android::AudioEffect* aec_;
android::AudioEffect* ns_;
#endif
// Audio status
uint16_t recording_delay_;

8
media/webrtc/trunk/webrtc/modules/audio_device/audio_device.gypi
View File

@ -52,6 +52,11 @@
'$(NSPR_CFLAGS)',
],
}],
['hardware_aec_ns==1', {
'defines': [
'WEBRTC_HARDWARE_AEC_NS',
],
}],
['OS=="linux" or include_alsa_audio==1 or include_pulse_audio==1', {
'include_dirs': [
'linux',
@ -81,7 +86,10 @@
['moz_widget_toolkit_gonk==1', {
'cflags_mozilla': [
'-I$(ANDROID_SOURCE)/frameworks/wilhelm/include',
'-I$(ANDROID_SOURCE)/frameworks/av/include',
'-I$(ANDROID_SOURCE)/system/media/wilhelm/include',
'-I$(ANDROID_SOURCE)/system/media/audio_effects/include',
'-I$(ANDROID_SOURCE)/frameworks/native/include',
],
'include_dirs': [
'android',

9
modules/libpref/init/all.js
View File

@ -353,11 +353,16 @@ pref("media.peerconnection.identity.timeout", 10000);
// kXxxUnchanged = 0, kXxxDefault = 1, and higher values are specific to each
// setting (for Xxx = Ec, Agc, or Ns). Defaults are all set to kXxxDefault here.
pref("media.peerconnection.turn.disable", false);
#if defined(MOZ_WEBRTC_HARDWARE_AEC_NS)
pref("media.getusermedia.aec_enabled", false);
pref("media.getusermedia.noise_enabled", false);
#else
pref("media.getusermedia.aec_enabled", true);
pref("media.getusermedia.noise_enabled", true);
#endif
pref("media.getusermedia.noise", 1);
pref("media.getusermedia.agc_enabled", false);
pref("media.getusermedia.agc", 1);
pref("media.getusermedia.noise_enabled", true);
pref("media.getusermedia.noise", 1);
// Adjustments for OS-specific input delay (lower bound)
// Adjustments for OS-specific AudioStream+cubeb+output delay (lower bound)
#if defined(XP_MACOSX)