Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 263 insertions, 0 deletions

diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc
new file mode 100644
index 0000000000..ba3212afc3
--- /dev/null
+++ b/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -0,0 +1,263 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_device/android/audio_record_jni.h"
+
+#include <android/log.h>
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/modules/audio_device/android/audio_common.h"
+
+#define TAG "AudioRecordJni"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+namespace webrtc {
+
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_record)
+    : audio_record_(audio_record.Pass()),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId(
+          "enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_agc_(native_reg->GetMethodId(
+          "enableBuiltInAGC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId(
+          "enableBuiltInNS", "(Z)Z")) {
+}
+
+AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
+
+int AudioRecordJni::JavaAudioRecord::InitRecording(
+    int sample_rate, int channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
+}
+
+bool AudioRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAGC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_agc_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      recording_(false),
+      audio_device_buffer_(nullptr) {
+  ALOGD("ctor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+      reinterpret_cast<void*>(
+          &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+      reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord",
+      native_methods, arraysize(native_methods));
+  j_audio_record_.reset(new JavaAudioRecord(
+      j_native_registration_.get(),
+      j_native_registration_->NewObject(
+          "<init>", "(Landroid/content/Context;J)V",
+          JVM::GetInstance()->context(), PointerTojlong(this))));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.DetachFromThread();
+}
+
+AudioRecordJni::~AudioRecordJni() {
+  ALOGD("~dtor%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  Terminate();
+}
+
+int32_t AudioRecordJni::Init() {
+  ALOGD("Init%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return 0;
+}
+
+int32_t AudioRecordJni::Terminate() {
+  ALOGD("Terminate%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  StopRecording();
+  return 0;
+}
+
+int32_t AudioRecordJni::InitRecording() {
+  ALOGD("InitRecording%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (frames_per_buffer < 0) {
+    ALOGE("InitRecording failed!");
+    return -1;
+  }
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * kBytesPerFrame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StartRecording() {
+  ALOGD("StartRecording%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (!j_audio_record_->StartRecording()) {
+    ALOGE("StartRecording failed!");
+    return -1;
+  }
+  recording_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StopRecording() {
+  ALOGD("StopRecording%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!j_audio_record_->StopRecording()) {
+    ALOGE("StopRecording failed!");
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.DetachFromThread();
+  initialized_ = false;
+  recording_ = false;
+  direct_buffer_address_= nullptr;
+  return 0;
+}
+
+void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  const int channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%d)", channels);
+  audio_device_buffer_->SetRecordingChannels(channels);
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  ALOGD("total_delay_in_milliseconds: %d", total_delay_in_milliseconds_);
+}
+
+int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
+  ALOGD("EnableBuiltInAEC%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
+  ALOGD("EnableBuiltInAGC%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInAGC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
+  ALOGD("EnableBuiltInNS%s", GetThreadInfo().c_str());
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
+}
+
+void JNICALL AudioRecordJni::CacheDirectBufferAddress(
+    JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*> (nativeAudioRecord);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioRecordJni::OnCacheDirectBufferAddress(
+    JNIEnv* env, jobject byte_buffer) {
+  ALOGD("OnCacheDirectBufferAddress");
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ =
+      env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  ALOGD("direct buffer capacity: %lld", capacity);
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+}
+
+void JNICALL AudioRecordJni::DataIsRecorded(
+  JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*> (nativeAudioRecord);
+  this_object->OnDataIsRecorded(length);
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordThread'.
+void AudioRecordJni::OnDataIsRecorded(int length) {
+  RTC_DCHECK(thread_checker_java_.CalledOnValidThread());
+  if (!audio_device_buffer_) {
+    ALOGE("AttachAudioBuffer has not been called!");
+    return;
+  }
+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
+                                          frames_per_buffer_);
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // |playDelayMs| parameter only. Components like the AEC only sees the sum
+  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_,
+                                   0,   // recDelayMs
+                                   0);  // clockDrift
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
+    ALOGE("AudioDeviceBuffer::DeliverRecordedData failed!");
+  }
+}
+
+}  // namespace webrtc