webrtc/examples/android/opensl_loopback/jni/opensl_runner.cc

/*
 *  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 <assert.h>
#include <jni.h>

#include "webrtc/examples/android/opensl_loopback/fake_audio_device_buffer.h"
#include "webrtc/modules/audio_device/android/audio_device_template.h"
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
#include "webrtc/modules/audio_device/android/opensles_input.h"
#include "webrtc/modules/audio_device/android/opensles_output.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"

// Java globals
static JavaVM* g_vm = NULL;
static jclass g_osr = NULL;

namespace webrtc {

template <class InputType, class OutputType>
class OpenSlRunnerTemplate {
 public:
  OpenSlRunnerTemplate()
      : output_(0),
        input_(0, &output_) {
    output_.AttachAudioBuffer(&audio_buffer_);
    if (output_.Init() != 0) {
      assert(false);
    }
    if (output_.InitPlayout() != 0) {
      assert(false);
    }
    input_.AttachAudioBuffer(&audio_buffer_);
    if (input_.Init() != 0) {
      assert(false);
    }
    if (input_.InitRecording() != 0) {
      assert(false);
    }
  }

  ~OpenSlRunnerTemplate() {}

  void StartPlayRecord() {
    output_.StartPlayout();
    input_.StartRecording();
  }

  void StopPlayRecord() {
    // There are large enough buffers to compensate for recording and playing
    // jitter such that the timing of stopping playing or recording should not
    // result in over or underrun.
    input_.StopRecording();
    output_.StopPlayout();
    audio_buffer_.ClearBuffer();
  }

 private:
  OutputType output_;
  InputType input_;
  FakeAudioDeviceBuffer audio_buffer_;
};

class OpenSlRunner
    : public OpenSlRunnerTemplate<OpenSlesInput, OpenSlesOutput> {
 public:
  // Global class implementing native code.
  static OpenSlRunner* g_runner;


  OpenSlRunner() {}
  virtual ~OpenSlRunner() {}

  static JNIEXPORT void JNICALL RegisterApplicationContext(
      JNIEnv* env,
      jobject obj,
      jobject context) {
    assert(!g_runner);  // Should only be called once.
    OpenSlesInput::SetAndroidAudioDeviceObjects(g_vm, env, context);
    OpenSlesOutput::SetAndroidAudioDeviceObjects(g_vm, env, context);
    g_runner = new OpenSlRunner();
  }

  static JNIEXPORT void JNICALL Start(JNIEnv * env, jobject) {
    g_runner->StartPlayRecord();
  }

  static JNIEXPORT void JNICALL Stop(JNIEnv * env, jobject) {
    g_runner->StopPlayRecord();
  }
};

OpenSlRunner* OpenSlRunner::g_runner = NULL;

}  // namespace webrtc

jint JNI_OnLoad(JavaVM* vm, void* reserved) {
  // Only called once.
  assert(!g_vm);
  JNIEnv* env;
  if (vm->GetEnv(reinterpret_cast<void**>(&env), JNI_VERSION_1_6) != JNI_OK) {
    return -1;
  }

  jclass local_osr = env->FindClass("org/webrtc/app/OpenSlRunner");
  assert(local_osr != NULL);
  g_osr = static_cast<jclass>(env->NewGlobalRef(local_osr));
  JNINativeMethod nativeFunctions[] = {
    {"RegisterApplicationContext", "(Landroid/content/Context;)V",
     reinterpret_cast<void*>(
         &webrtc::OpenSlRunner::RegisterApplicationContext)},
    {"Start", "()V", reinterpret_cast<void*>(&webrtc::OpenSlRunner::Start)},
    {"Stop",  "()V", reinterpret_cast<void*>(&webrtc::OpenSlRunner::Stop)}
  };
  int ret_val = env->RegisterNatives(g_osr, nativeFunctions, 3);
  if (ret_val != 0) {
    assert(false);
  }
  g_vm = vm;
  return JNI_VERSION_1_6;
}