blob: a3aa85565af6e80baef4f029775ef651b941b6d2 [file] [log] [blame]
/*
* 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 "modules/audio_device/android/audio_record_jni.h"
#include <string>
#include <utility>
#include "modules/audio_device/android/audio_common.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
// Scoped class which logs its time of life as a UMA statistic. It generates
// a histogram which measures the time it takes for a method/scope to execute.
class ScopedHistogramTimer {
public:
explicit ScopedHistogramTimer(const std::string& name)
: histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
~ScopedHistogramTimer() {
const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
RTC_LOG(INFO) << histogram_name_ << ": " << life_time_ms;
}
private:
const std::string histogram_name_;
int64_t start_time_ms_;
};
} // namespace
// AudioRecordJni::JavaAudioRecord implementation.
AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
NativeRegistration* native_reg,
std::unique_ptr<GlobalRef> audio_record)
: audio_record_(std::move(audio_record)),
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_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
int AudioRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
size_t 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::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) {
RTC_LOG(INFO) << "ctor";
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>", "(J)V", 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_.Detach();
}
AudioRecordJni::~AudioRecordJni() {
RTC_LOG(INFO) << "dtor";
RTC_DCHECK(thread_checker_.IsCurrent());
Terminate();
}
int32_t AudioRecordJni::Init() {
RTC_LOG(INFO) << "Init";
RTC_DCHECK(thread_checker_.IsCurrent());
return 0;
}
int32_t AudioRecordJni::Terminate() {
RTC_LOG(INFO) << "Terminate";
RTC_DCHECK(thread_checker_.IsCurrent());
StopRecording();
return 0;
}
int32_t AudioRecordJni::InitRecording() {
RTC_LOG(INFO) << "InitRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!recording_);
ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
int frames_per_buffer = j_audio_record_->InitRecording(
audio_parameters_.sample_rate(), audio_parameters_.channels());
if (frames_per_buffer < 0) {
direct_buffer_address_ = nullptr;
RTC_LOG(LS_ERROR) << "InitRecording failed";
return -1;
}
frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
frames_per_buffer_ * bytes_per_frame);
RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
initialized_ = true;
return 0;
}
int32_t AudioRecordJni::StartRecording() {
RTC_LOG(INFO) << "StartRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!recording_);
if (!initialized_) {
RTC_DLOG(LS_WARNING)
<< "Recording can not start since InitRecording must succeed first";
return 0;
}
ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
if (!j_audio_record_->StartRecording()) {
RTC_LOG(LS_ERROR) << "StartRecording failed";
return -1;
}
recording_ = true;
return 0;
}
int32_t AudioRecordJni::StopRecording() {
RTC_LOG(INFO) << "StopRecording";
RTC_DCHECK(thread_checker_.IsCurrent());
if (!initialized_ || !recording_) {
return 0;
}
if (!j_audio_record_->StopRecording()) {
RTC_LOG(LS_ERROR) << "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_.Detach();
initialized_ = false;
recording_ = false;
direct_buffer_address_ = nullptr;
return 0;
}
void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
RTC_LOG(INFO) << "AttachAudioBuffer";
RTC_DCHECK(thread_checker_.IsCurrent());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
RTC_LOG(INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
audio_device_buffer_->SetRecordingChannels(channels);
total_delay_in_milliseconds_ =
audio_manager_->GetDelayEstimateInMilliseconds();
RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
RTC_LOG(INFO) << "total_delay_in_milliseconds: "
<< total_delay_in_milliseconds_;
}
int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
RTC_LOG(INFO) << "EnableBuiltInAEC(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
}
int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
// TODO(henrika): possibly remove when no longer used by any client.
RTC_CHECK_NOTREACHED();
}
int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
RTC_LOG(INFO) << "EnableBuiltInNS(" << enable << ")";
RTC_DCHECK(thread_checker_.IsCurrent());
return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
}
JNI_FUNCTION_ALIGN
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) {
RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
RTC_DCHECK(thread_checker_.IsCurrent());
RTC_DCHECK(!direct_buffer_address_);
direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
}
JNI_FUNCTION_ALIGN
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_.IsCurrent());
if (!audio_device_buffer_) {
RTC_LOG(LS_ERROR) << "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);
if (audio_device_buffer_->DeliverRecordedData() == -1) {
RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
}
}
} // namespace webrtc