blob: 0ffb2594a570539c769218cd5c17b4d1a7c937e3 [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 "webrtc/modules/audio_device/android/opensles_player.h"
#include <android/log.h>
#include "webrtc/modules/audio_device/android/audio_common.h"
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/fine_audio_buffer.h"
#include "webrtc/rtc_base/array_view.h"
#include "webrtc/rtc_base/arraysize.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/format_macros.h"
#include "webrtc/rtc_base/timeutils.h"
#define TAG "OpenSLESPlayer"
#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__)
#define RETURN_ON_ERROR(op, ...) \
do { \
SLresult err = (op); \
if (err != SL_RESULT_SUCCESS) { \
ALOGE("%s failed: %s", #op, GetSLErrorString(err)); \
return __VA_ARGS__; \
} \
} while (0)
namespace webrtc {
OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
: audio_manager_(audio_manager),
audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
audio_device_buffer_(nullptr),
initialized_(false),
playing_(false),
buffer_index_(0),
engine_(nullptr),
player_(nullptr),
simple_buffer_queue_(nullptr),
volume_(nullptr),
last_play_time_(0) {
ALOGD("ctor%s", GetThreadInfo().c_str());
// Use native audio output parameters provided by the audio manager and
// define the PCM format structure.
pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
audio_parameters_.sample_rate(),
audio_parameters_.bits_per_sample());
// Detach from this thread since we want to use the checker to verify calls
// from the internal audio thread.
thread_checker_opensles_.DetachFromThread();
}
OpenSLESPlayer::~OpenSLESPlayer() {
ALOGD("dtor%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
Terminate();
DestroyAudioPlayer();
DestroyMix();
engine_ = nullptr;
RTC_DCHECK(!engine_);
RTC_DCHECK(!output_mix_.Get());
RTC_DCHECK(!player_);
RTC_DCHECK(!simple_buffer_queue_);
RTC_DCHECK(!volume_);
}
int OpenSLESPlayer::Init() {
ALOGD("Init%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return 0;
}
int OpenSLESPlayer::Terminate() {
ALOGD("Terminate%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
StopPlayout();
return 0;
}
int OpenSLESPlayer::InitPlayout() {
ALOGD("InitPlayout%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!initialized_);
RTC_DCHECK(!playing_);
if (!ObtainEngineInterface()) {
ALOGE("Failed to obtain SL Engine interface");
return -1;
}
CreateMix();
initialized_ = true;
buffer_index_ = 0;
return 0;
}
int OpenSLESPlayer::StartPlayout() {
ALOGD("StartPlayout%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(initialized_);
RTC_DCHECK(!playing_);
if (fine_audio_buffer_) {
fine_audio_buffer_->ResetPlayout();
}
// The number of lower latency audio players is limited, hence we create the
// audio player in Start() and destroy it in Stop().
CreateAudioPlayer();
// Fill up audio buffers to avoid initial glitch and to ensure that playback
// starts when mode is later changed to SL_PLAYSTATE_PLAYING.
// TODO(henrika): we can save some delay by only making one call to
// EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
last_play_time_ = rtc::Time();
for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
EnqueuePlayoutData(true);
}
// Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
// For a player object, when the object is in the SL_PLAYSTATE_PLAYING
// state, adding buffers will implicitly start playback.
RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
RTC_DCHECK(playing_);
return 0;
}
int OpenSLESPlayer::StopPlayout() {
ALOGD("StopPlayout%s", GetThreadInfo().c_str());
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (!initialized_ || !playing_) {
return 0;
}
// Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
// Clear the buffer queue to flush out any remaining data.
RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
#if RTC_DCHECK_IS_ON
// Verify that the buffer queue is in fact cleared as it should.
SLAndroidSimpleBufferQueueState buffer_queue_state;
(*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
RTC_DCHECK_EQ(0, buffer_queue_state.count);
RTC_DCHECK_EQ(0, buffer_queue_state.index);
#endif
// The number of lower latency audio players is limited, hence we create the
// audio player in Start() and destroy it in Stop().
DestroyAudioPlayer();
thread_checker_opensles_.DetachFromThread();
initialized_ = false;
playing_ = false;
return 0;
}
int OpenSLESPlayer::SpeakerVolumeIsAvailable(bool& available) {
available = false;
return 0;
}
int OpenSLESPlayer::MaxSpeakerVolume(uint32_t& maxVolume) const {
return -1;
}
int OpenSLESPlayer::MinSpeakerVolume(uint32_t& minVolume) const {
return -1;
}
int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
return -1;
}
int OpenSLESPlayer::SpeakerVolume(uint32_t& volume) const {
return -1;
}
void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
ALOGD("AttachAudioBuffer");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
audio_device_buffer_ = audioBuffer;
const int sample_rate_hz = audio_parameters_.sample_rate();
ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
const size_t channels = audio_parameters_.channels();
ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
audio_device_buffer_->SetPlayoutChannels(channels);
RTC_CHECK(audio_device_buffer_);
AllocateDataBuffers();
}
void OpenSLESPlayer::AllocateDataBuffers() {
ALOGD("AllocateDataBuffers");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(!simple_buffer_queue_);
RTC_CHECK(audio_device_buffer_);
// Create a modified audio buffer class which allows us to ask for any number
// of samples (and not only multiple of 10ms) to match the native OpenSL ES
// buffer size. The native buffer size corresponds to the
// PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
// frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
// recommended to construct audio buffers so that they contain an exact
// multiple of this number. If so, callbacks will occur at regular intervals,
// which reduces jitter.
const size_t buffer_size_in_bytes = audio_parameters_.GetBytesPerBuffer();
ALOGD("native buffer size: %" PRIuS, buffer_size_in_bytes);
ALOGD("native buffer size in ms: %.2f",
audio_parameters_.GetBufferSizeInMilliseconds());
fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_,
audio_parameters_.sample_rate(),
2 * buffer_size_in_bytes));
// Allocated memory for audio buffers.
for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
audio_buffers_[i].reset(new SLint8[buffer_size_in_bytes]);
}
}
bool OpenSLESPlayer::ObtainEngineInterface() {
ALOGD("ObtainEngineInterface");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (engine_)
return true;
// Get access to (or create if not already existing) the global OpenSL Engine
// object.
SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
if (engine_object == nullptr) {
ALOGE("Failed to access the global OpenSL engine");
return false;
}
// Get the SL Engine Interface which is implicit.
RETURN_ON_ERROR(
(*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine_),
false);
return true;
}
bool OpenSLESPlayer::CreateMix() {
ALOGD("CreateMix");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(engine_);
if (output_mix_.Get())
return true;
// Create the ouput mix on the engine object. No interfaces will be used.
RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
nullptr, nullptr),
false);
RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
false);
return true;
}
void OpenSLESPlayer::DestroyMix() {
ALOGD("DestroyMix");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (!output_mix_.Get())
return;
output_mix_.Reset();
}
bool OpenSLESPlayer::CreateAudioPlayer() {
ALOGD("CreateAudioPlayer");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
RTC_DCHECK(output_mix_.Get());
if (player_object_.Get())
return true;
RTC_DCHECK(!player_);
RTC_DCHECK(!simple_buffer_queue_);
RTC_DCHECK(!volume_);
// source: Android Simple Buffer Queue Data Locator is source.
SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
// sink: OutputMix-based data is sink.
SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
output_mix_.Get()};
SLDataSink audio_sink = {&locator_output_mix, nullptr};
// Define interfaces that we indend to use and realize.
const SLInterfaceID interface_ids[] = {
SL_IID_ANDROIDCONFIGURATION, SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
const SLboolean interface_required[] = {
SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
// Create the audio player on the engine interface.
RETURN_ON_ERROR(
(*engine_)->CreateAudioPlayer(
engine_, player_object_.Receive(), &audio_source, &audio_sink,
arraysize(interface_ids), interface_ids, interface_required),
false);
// Use the Android configuration interface to set platform-specific
// parameters. Should be done before player is realized.
SLAndroidConfigurationItf player_config;
RETURN_ON_ERROR(
player_object_->GetInterface(player_object_.Get(),
SL_IID_ANDROIDCONFIGURATION, &player_config),
false);
// Set audio player configuration to SL_ANDROID_STREAM_VOICE which
// corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
RETURN_ON_ERROR(
(*player_config)
->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
&stream_type, sizeof(SLint32)),
false);
// Realize the audio player object after configuration has been set.
RETURN_ON_ERROR(
player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
// Get the SLPlayItf interface on the audio player.
RETURN_ON_ERROR(
player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY, &player_),
false);
// Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
RETURN_ON_ERROR(
player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE,
&simple_buffer_queue_),
false);
// Register callback method for the Android Simple Buffer Queue interface.
// This method will be called when the native audio layer needs audio data.
RETURN_ON_ERROR((*simple_buffer_queue_)
->RegisterCallback(simple_buffer_queue_,
SimpleBufferQueueCallback, this),
false);
// Get the SLVolumeItf interface on the audio player.
RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
SL_IID_VOLUME, &volume_),
false);
// TODO(henrika): might not be required to set volume to max here since it
// seems to be default on most devices. Might be required for unit tests.
// RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
return true;
}
void OpenSLESPlayer::DestroyAudioPlayer() {
ALOGD("DestroyAudioPlayer");
RTC_DCHECK(thread_checker_.CalledOnValidThread());
if (!player_object_.Get())
return;
(*simple_buffer_queue_)
->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
player_object_.Reset();
player_ = nullptr;
simple_buffer_queue_ = nullptr;
volume_ = nullptr;
}
// static
void OpenSLESPlayer::SimpleBufferQueueCallback(
SLAndroidSimpleBufferQueueItf caller,
void* context) {
OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
stream->FillBufferQueue();
}
void OpenSLESPlayer::FillBufferQueue() {
RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
SLuint32 state = GetPlayState();
if (state != SL_PLAYSTATE_PLAYING) {
ALOGW("Buffer callback in non-playing state!");
return;
}
EnqueuePlayoutData(false);
}
void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
// Check delta time between two successive callbacks and provide a warning
// if it becomes very large.
// TODO(henrika): using 150ms as upper limit but this value is rather random.
const uint32_t current_time = rtc::Time();
const uint32_t diff = current_time - last_play_time_;
if (diff > 150) {
ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
}
last_play_time_ = current_time;
SLint8* audio_ptr = audio_buffers_[buffer_index_].get();
if (silence) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
// Avoid aquiring real audio data from WebRTC and fill the buffer with
// zeros instead. Used to prime the buffer with silence and to avoid asking
// for audio data from two different threads.
memset(audio_ptr, 0, audio_parameters_.GetBytesPerBuffer());
} else {
RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
// Read audio data from the WebRTC source using the FineAudioBuffer object
// to adjust for differences in buffer size between WebRTC (10ms) and native
// OpenSL ES.
fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<SLint8>(
audio_ptr, audio_parameters_.GetBytesPerBuffer()));
}
// Enqueue the decoded audio buffer for playback.
SLresult err = (*simple_buffer_queue_)
->Enqueue(simple_buffer_queue_, audio_ptr,
audio_parameters_.GetBytesPerBuffer());
if (SL_RESULT_SUCCESS != err) {
ALOGE("Enqueue failed: %d", err);
}
buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
}
SLuint32 OpenSLESPlayer::GetPlayState() const {
RTC_DCHECK(player_);
SLuint32 state;
SLresult err = (*player_)->GetPlayState(player_, &state);
if (SL_RESULT_SUCCESS != err) {
ALOGE("GetPlayState failed: %d", err);
}
return state;
}
} // namespace webrtc