blob: d6fcd464ac24df0127147c2a2761cece91c68fc5 [file] [log] [blame]
* Copyright (c) 2015 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 <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
#include <SLES/OpenSLES_AndroidConfiguration.h>
#include "webrtc/modules/audio_device/android/audio_common.h"
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/android/opensles_common.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/utility/include/helpers_android.h"
#include "webrtc/rtc_base/thread_checker.h"
namespace webrtc {
class FineAudioBuffer;
// Implements 16-bit mono PCM audio output support for Android using the
// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
// An instance must be created and destroyed on one and the same thread.
// All public methods must also be called on the same thread. A thread checker
// will RTC_DCHECK if any method is called on an invalid thread. Decoded audio
// buffers are requested on a dedicated internal thread managed by the OpenSL
// ES layer.
// The existing design forces the user to call InitPlayout() after Stoplayout()
// to be able to call StartPlayout() again. This is inline with how the Java-
// based implementation works.
// OpenSL ES is a native C API which have no Dalvik-related overhead such as
// garbage collection pauses and it supports reduced audio output latency.
// If the device doesn't claim this feature but supports API level 9 (Android
// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
// the output latency may be higher.
class OpenSLESPlayer {
// Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
// required for lower latency. Beginning with API level 18 (Android 4.3), a
// buffer count of 1 is sufficient for lower latency. In addition, the buffer
// size and sample rate must be compatible with the device's native output
// configuration provided via the audio manager at construction.
// TODO(henrika): perhaps set this value dynamically based on OS version.
static const int kNumOfOpenSLESBuffers = 2;
explicit OpenSLESPlayer(AudioManager* audio_manager);
int Init();
int Terminate();
int InitPlayout();
bool PlayoutIsInitialized() const { return initialized_; }
int StartPlayout();
int StopPlayout();
bool Playing() const { return playing_; }
int SpeakerVolumeIsAvailable(bool& available);
int SetSpeakerVolume(uint32_t volume);
int SpeakerVolume(uint32_t& volume) const;
int MaxSpeakerVolume(uint32_t& maxVolume) const;
int MinSpeakerVolume(uint32_t& minVolume) const;
void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
// These callback methods are called when data is required for playout.
// They are both called from an internal "OpenSL ES thread" which is not
// attached to the Dalvik VM.
static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
void* context);
void FillBufferQueue();
// Reads audio data in PCM format using the AudioDeviceBuffer.
// Can be called both on the main thread (during Start()) and from the
// internal audio thread while output streaming is active.
// If the |silence| flag is set, the audio is filled with zeros instead of
// asking the WebRTC layer for real audio data. This procedure is also known
// as audio priming.
void EnqueuePlayoutData(bool silence);
// Allocate memory for audio buffers which will be used to render audio
// via the SLAndroidSimpleBufferQueueItf interface.
void AllocateDataBuffers();
// Obtaines the SL Engine Interface from the existing global Engine object.
// The interface exposes creation methods of all the OpenSL ES object types.
// This method defines the |engine_| member variable.
bool ObtainEngineInterface();
// Creates/destroys the output mix object.
bool CreateMix();
void DestroyMix();
// Creates/destroys the audio player and the simple-buffer object.
// Also creates the volume object.
bool CreateAudioPlayer();
void DestroyAudioPlayer();
SLuint32 GetPlayState() const;
// Ensures that methods are called from the same thread as this object is
// created on.
rtc::ThreadChecker thread_checker_;
// Stores thread ID in first call to SimpleBufferQueueCallback() from internal
// non-application thread which is not attached to the Dalvik JVM.
// Detached during construction of this object.
rtc::ThreadChecker thread_checker_opensles_;
// Raw pointer to the audio manager injected at construction. Used to cache
// audio parameters and to access the global SL engine object needed by the
// ObtainEngineInterface() method. The audio manager outlives any instance of
// this class.
AudioManager* audio_manager_;
// Contains audio parameters provided to this class at construction by the
// AudioManager.
const AudioParameters audio_parameters_;
// Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
// AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
AudioDeviceBuffer* audio_device_buffer_;
bool initialized_;
bool playing_;
// PCM-type format definition.
// TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
// 32-bit float representation is needed.
SLDataFormat_PCM pcm_format_;
// Queue of audio buffers to be used by the player object for rendering
// audio. They will be used in a Round-robin way and the size of each buffer
// is given by FineAudioBuffer::RequiredBufferSizeBytes().
std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
// in chunks of 10ms. It then allows for this data to be pulled in
// a finer or coarser granularity. I.e. interacting with this class instead
// of directly with the AudioDeviceBuffer one can ask for any number of
// audio data samples.
// Example: native buffer size can be 192 audio frames at 48kHz sample rate.
// WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 192
// in each callback (one every 4th ms). This class can then ask for 192 and
// the FineAudioBuffer will ask WebRTC for new data approximately only every
// second callback and also cache non-utilized audio.
std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
// Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
// Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
int buffer_index_;
// This interface exposes creation methods for all the OpenSL ES object types.
// It is the OpenSL ES API entry point.
SLEngineItf engine_;
// Output mix object to be used by the player object.
webrtc::ScopedSLObjectItf output_mix_;
// The audio player media object plays out audio to the speakers. It also
// supports volume control.
webrtc::ScopedSLObjectItf player_object_;
// This interface is supported on the audio player and it controls the state
// of the audio player.
SLPlayItf player_;
// The Android Simple Buffer Queue interface is supported on the audio player
// and it provides methods to send audio data from the source to the audio
// player for rendering.
SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
// This interface exposes controls for manipulating the object’s audio volume
// properties. This interface is supported on the Audio Player object.
SLVolumeItf volume_;
// Last time the OpenSL ES layer asked for audio data to play out.
uint32_t last_play_time_;
} // namespace webrtc