| /* |
| * 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_track_jni.h" |
| #include "modules/audio_device/android/audio_manager.h" |
| |
| #include <utility> |
| |
| #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" |
| |
| namespace webrtc { |
| |
| // AudioTrackJni::JavaAudioTrack implementation. |
| AudioTrackJni::JavaAudioTrack::JavaAudioTrack( |
| NativeRegistration* native_reg, |
| std::unique_ptr<GlobalRef> audio_track) |
| : audio_track_(std::move(audio_track)), |
| init_playout_(native_reg->GetMethodId("initPlayout", "(II)Z")), |
| start_playout_(native_reg->GetMethodId("startPlayout", "()Z")), |
| stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")), |
| set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")), |
| get_stream_max_volume_( |
| native_reg->GetMethodId("getStreamMaxVolume", "()I")), |
| get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {} |
| |
| AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {} |
| |
| bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) { |
| return audio_track_->CallBooleanMethod(init_playout_, sample_rate, channels); |
| } |
| |
| bool AudioTrackJni::JavaAudioTrack::StartPlayout() { |
| return audio_track_->CallBooleanMethod(start_playout_); |
| } |
| |
| bool AudioTrackJni::JavaAudioTrack::StopPlayout() { |
| return audio_track_->CallBooleanMethod(stop_playout_); |
| } |
| |
| bool AudioTrackJni::JavaAudioTrack::SetStreamVolume(int volume) { |
| return audio_track_->CallBooleanMethod(set_stream_volume_, volume); |
| } |
| |
| int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() { |
| return audio_track_->CallIntMethod(get_stream_max_volume_); |
| } |
| |
| int AudioTrackJni::JavaAudioTrack::GetStreamVolume() { |
| return audio_track_->CallIntMethod(get_stream_volume_); |
| } |
| |
| // TODO(henrika): possible extend usage of AudioManager and add it as member. |
| AudioTrackJni::AudioTrackJni(AudioManager* audio_manager) |
| : j_environment_(JVM::GetInstance()->environment()), |
| audio_parameters_(audio_manager->GetPlayoutAudioParameters()), |
| direct_buffer_address_(nullptr), |
| direct_buffer_capacity_in_bytes_(0), |
| frames_per_buffer_(0), |
| initialized_(false), |
| playing_(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::AudioTrackJni::CacheDirectBufferAddress)}, |
| {"nativeGetPlayoutData", "(IJ)V", |
| reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}}; |
| j_native_registration_ = j_environment_->RegisterNatives( |
| "org/webrtc/voiceengine/WebRtcAudioTrack", native_methods, |
| arraysize(native_methods)); |
| j_audio_track_.reset( |
| new JavaAudioTrack(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_.DetachFromThread(); |
| } |
| |
| AudioTrackJni::~AudioTrackJni() { |
| RTC_LOG(INFO) << "dtor"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| Terminate(); |
| } |
| |
| int32_t AudioTrackJni::Init() { |
| RTC_LOG(INFO) << "Init"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return 0; |
| } |
| |
| int32_t AudioTrackJni::Terminate() { |
| RTC_LOG(INFO) << "Terminate"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| StopPlayout(); |
| return 0; |
| } |
| |
| int32_t AudioTrackJni::InitPlayout() { |
| RTC_LOG(INFO) << "InitPlayout"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| RTC_DCHECK(!initialized_); |
| RTC_DCHECK(!playing_); |
| if (!j_audio_track_->InitPlayout(audio_parameters_.sample_rate(), |
| audio_parameters_.channels())) { |
| RTC_LOG(LS_ERROR) << "InitPlayout failed"; |
| return -1; |
| } |
| initialized_ = true; |
| return 0; |
| } |
| |
| int32_t AudioTrackJni::StartPlayout() { |
| RTC_LOG(INFO) << "StartPlayout"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| RTC_DCHECK(!playing_); |
| if (!initialized_) { |
| RTC_DLOG(LS_WARNING) |
| << "Playout can not start since InitPlayout must succeed first"; |
| return 0; |
| } |
| if (!j_audio_track_->StartPlayout()) { |
| RTC_LOG(LS_ERROR) << "StartPlayout failed"; |
| return -1; |
| } |
| playing_ = true; |
| return 0; |
| } |
| |
| int32_t AudioTrackJni::StopPlayout() { |
| RTC_LOG(INFO) << "StopPlayout"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| if (!initialized_ || !playing_) { |
| return 0; |
| } |
| if (!j_audio_track_->StopPlayout()) { |
| RTC_LOG(LS_ERROR) << "StopPlayout 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; |
| playing_ = false; |
| direct_buffer_address_ = nullptr; |
| return 0; |
| } |
| |
| int AudioTrackJni::SpeakerVolumeIsAvailable(bool& available) { |
| available = true; |
| return 0; |
| } |
| |
| int AudioTrackJni::SetSpeakerVolume(uint32_t volume) { |
| RTC_LOG(INFO) << "SetSpeakerVolume(" << volume << ")"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return j_audio_track_->SetStreamVolume(volume) ? 0 : -1; |
| } |
| |
| int AudioTrackJni::MaxSpeakerVolume(uint32_t& max_volume) const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| max_volume = j_audio_track_->GetStreamMaxVolume(); |
| return 0; |
| } |
| |
| int AudioTrackJni::MinSpeakerVolume(uint32_t& min_volume) const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| min_volume = 0; |
| return 0; |
| } |
| |
| int AudioTrackJni::SpeakerVolume(uint32_t& volume) const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| volume = j_audio_track_->GetStreamVolume(); |
| RTC_LOG(INFO) << "SpeakerVolume: " << volume; |
| return 0; |
| } |
| |
| // TODO(henrika): possibly add stereo support. |
| void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) { |
| RTC_LOG(INFO) << "AttachAudioBuffer"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| audio_device_buffer_ = audioBuffer; |
| const int sample_rate_hz = audio_parameters_.sample_rate(); |
| RTC_LOG(INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")"; |
| audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz); |
| const size_t channels = audio_parameters_.channels(); |
| RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")"; |
| audio_device_buffer_->SetPlayoutChannels(channels); |
| } |
| |
| JNI_FUNCTION_ALIGN |
| void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env, |
| jobject obj, |
| jobject byte_buffer, |
| jlong nativeAudioTrack) { |
| webrtc::AudioTrackJni* this_object = |
| reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack); |
| this_object->OnCacheDirectBufferAddress(env, byte_buffer); |
| } |
| |
| void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env, |
| jobject byte_buffer) { |
| RTC_LOG(INFO) << "OnCacheDirectBufferAddress"; |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| 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); |
| const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t); |
| frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame; |
| RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_; |
| } |
| |
| JNI_FUNCTION_ALIGN |
| void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env, |
| jobject obj, |
| jint length, |
| jlong nativeAudioTrack) { |
| webrtc::AudioTrackJni* this_object = |
| reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack); |
| this_object->OnGetPlayoutData(static_cast<size_t>(length)); |
| } |
| |
| // This method is called on a high-priority thread from Java. The name of |
| // the thread is 'AudioRecordTrack'. |
| void AudioTrackJni::OnGetPlayoutData(size_t length) { |
| RTC_DCHECK(thread_checker_java_.CalledOnValidThread()); |
| const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t); |
| RTC_DCHECK_EQ(frames_per_buffer_, length / bytes_per_frame); |
| if (!audio_device_buffer_) { |
| RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called"; |
| return; |
| } |
| // Pull decoded data (in 16-bit PCM format) from jitter buffer. |
| int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_); |
| if (samples <= 0) { |
| RTC_LOG(LS_ERROR) << "AudioDeviceBuffer::RequestPlayoutData failed"; |
| return; |
| } |
| RTC_DCHECK_EQ(samples, frames_per_buffer_); |
| // Copy decoded data into common byte buffer to ensure that it can be |
| // written to the Java based audio track. |
| samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_); |
| RTC_DCHECK_EQ(length, bytes_per_frame * samples); |
| } |
| |
| } // namespace webrtc |