Refactor OpenSL audio driver
Message: I want to start this review, the basica framework is almost done. Description: This implementation is similar to current one, but 1. followed the design doc at https://docs.google.com/a/google.com/document/d/1g5q2SVtkFPl2OSjvSF3eeLb_S7sCnbINxiBUES6XLCM/edit which uses two threads, playout thread and recording thread, uses large audio buffer, etc. 2. google code style. What are missing in this cl, 1. a better way to control schedule/thread priority. 2. java/jni interface to better support what cannot be done in OpenSL. Please take a review, thanks. Review URL: https://webrtc-codereview.appspot.com/902004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3040 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
leozwang@webrtc.org
parent
737ed3bfa5
commit
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@ -8,14 +8,10 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_OPENSLES_ANDROID_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_OPENSLES_ANDROID_H
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#ifndef SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_
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#define SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_
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#include "audio_device_generic.h"
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#include "critical_section_wrapper.h"
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#include <jni.h> // For accessing AudioDeviceAndroid.java
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#include <queue>
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#include <jni.h>
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#include <stdio.h>
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#include <stdlib.h>
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@ -23,41 +19,55 @@
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#include <SLES/OpenSLES_Android.h>
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#include <SLES/OpenSLES_AndroidConfiguration.h>
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namespace webrtc
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{
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#include <queue>
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#include "modules/audio_device/audio_device_generic.h"
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#include "system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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class EventWrapper;
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const WebRtc_UWord32 N_MAX_INTERFACES = 3;
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const WebRtc_UWord32 N_MAX_OUTPUT_DEVICES = 6;
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const WebRtc_UWord32 N_MAX_INPUT_DEVICES = 3;
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const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 16000;//44000; // Default fs
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const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000;//44000; // Default fs
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const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 16000; // Default fs
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const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000; // Default fs
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const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
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const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout
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const WebRtc_UWord32 N_REC_CHANNELS = 1;
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const WebRtc_UWord32 N_PLAY_CHANNELS = 1;
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const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
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const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480;
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const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = 480;
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const WebRtc_UWord32 REC_MAX_TEMP_BUF_SIZE_PER_10ms =
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N_REC_CHANNELS * REC_BUF_SIZE_IN_SAMPLES * sizeof(int16_t);
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const WebRtc_UWord32 PLAY_MAX_TEMP_BUF_SIZE_PER_10ms =
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N_PLAY_CHANNELS * PLAY_BUF_SIZE_IN_SAMPLES * sizeof(int16_t);
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// Number of the buffers in playout queue
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const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 2;
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const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 8;
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// Number of buffers in recording queue
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const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 2;
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// Number of 10 ms recording blocks in rec buffer
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const WebRtc_UWord16 N_REC_BUFFERS = 20;
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// TODO(xian): Reduce the numbers of buffers to improve the latency.
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const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 8;
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// Some values returned from getMinBufferSize
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// (Nexus S playout 72ms, recording 64ms)
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// (Galaxy, 167ms, 44ms)
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// (Nexus 7, 72ms, 48ms)
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// (Xoom 92ms, 40ms)
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class ThreadWrapper;
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class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric
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{
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public:
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AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id);
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class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric {
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public:
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explicit AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id);
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~AudioDeviceAndroidOpenSLES();
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// Retrieve the currently utilized audio layer
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virtual WebRtc_Word32
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ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
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ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; // NOLINT
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// Main initializaton and termination
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virtual WebRtc_Word32 Init();
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@ -85,10 +95,10 @@ public:
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SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
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// Audio transport initialization
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virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
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virtual WebRtc_Word32 PlayoutIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 InitPlayout();
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virtual bool PlayoutIsInitialized() const;
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virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
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virtual WebRtc_Word32 RecordingIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 InitRecording();
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virtual bool RecordingIsInitialized() const;
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@ -107,69 +117,82 @@ public:
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// Volume control based on the Windows Wave API (Windows only)
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virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
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WebRtc_UWord16 volumeRight);
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virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
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WebRtc_UWord16& volumeRight) const;
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virtual WebRtc_Word32 WaveOutVolume(
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WebRtc_UWord16& volumeLeft, // NOLINT
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WebRtc_UWord16& volumeRight) const; // NOLINT
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// Audio mixer initialization
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virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
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virtual WebRtc_Word32 SpeakerIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 InitSpeaker();
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virtual bool SpeakerIsInitialized() const;
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SLPlayItf playItf;
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virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
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virtual WebRtc_Word32 MicrophoneIsAvailable(
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bool& available);
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virtual WebRtc_Word32 InitMicrophone();
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virtual bool MicrophoneIsInitialized() const;
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// Speaker volume controls
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virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
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virtual WebRtc_Word32 SpeakerVolumeIsAvailable(
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bool& available); // NOLINT
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virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
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virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
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virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
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virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
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virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
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virtual WebRtc_Word32 SpeakerVolume(
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WebRtc_UWord32& volume) const; // NOLINT
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virtual WebRtc_Word32 MaxSpeakerVolume(
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WebRtc_UWord32& maxVolume) const; // NOLINT
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virtual WebRtc_Word32 MinSpeakerVolume(
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WebRtc_UWord32& minVolume) const; // NOLINT
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virtual WebRtc_Word32 SpeakerVolumeStepSize(
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WebRtc_UWord16& stepSize) const; // NOLINT
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// Microphone volume controls
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virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
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virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(
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bool& available); // NOLINT
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virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
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virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
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virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
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virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
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virtual WebRtc_Word32 MicrophoneVolume(
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WebRtc_UWord32& volume) const; // NOLINT
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virtual WebRtc_Word32 MaxMicrophoneVolume(
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WebRtc_UWord32& maxVolume) const; // NOLINT
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virtual WebRtc_Word32 MinMicrophoneVolume(
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WebRtc_UWord32& minVolume) const; // NOLINT
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virtual WebRtc_Word32
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MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
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MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const; // NOLINT
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// Speaker mute control
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virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
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virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 SetSpeakerMute(bool enable);
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virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
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virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; // NOLINT
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// Microphone mute control
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virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
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virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
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virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
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virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const; // NOLINT
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// Microphone boost control
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virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
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virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
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virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
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virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const; // NOLINT
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// Stereo support
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virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
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virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 SetStereoPlayout(bool enable);
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virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
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virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
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virtual WebRtc_Word32 StereoPlayout(bool& enabled) const; // NOLINT
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virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available); // NOLINT
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virtual WebRtc_Word32 SetStereoRecording(bool enable);
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virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
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virtual WebRtc_Word32 StereoRecording(bool& enabled) const; // NOLINT
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// Delay information and control
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virtual WebRtc_Word32
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SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
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WebRtc_UWord16 sizeMS);
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virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
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virtual WebRtc_Word32 PlayoutBuffer(
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AudioDeviceModule::BufferType& type, // NOLINT
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WebRtc_UWord16& sizeMS) const;
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virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
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virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
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virtual WebRtc_Word32 PlayoutDelay(
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WebRtc_UWord16& delayMS) const; // NOLINT
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virtual WebRtc_Word32 RecordingDelay(
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WebRtc_UWord16& delayMS) const; // NOLINT
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// CPU load
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virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
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virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const; // NOLINT
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// Error and warning information
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virtual bool PlayoutWarning() const;
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@ -186,27 +209,25 @@ public:
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// Speaker audio routing
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virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
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virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
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virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const; // NOLINT
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private:
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private:
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// Lock
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void Lock()
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{
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_critSect.Enter();
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void Lock() {
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crit_sect_.Enter();
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};
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void UnLock()
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{
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_critSect.Leave();
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void UnLock() {
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crit_sect_.Leave();
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};
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static void PlayerSimpleBufferQueueCallback(
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SLAndroidSimpleBufferQueueItf queueItf,
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void *pContext);
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void PlayerSimpleBufferQueueCallbackHandler(
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SLAndroidSimpleBufferQueueItf queueItf);
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static void RecorderSimpleBufferQueueCallback(
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SLAndroidSimpleBufferQueueItf queueItf,
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void *pContext);
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void PlayerSimpleBufferQueueCallbackHandler(
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SLAndroidSimpleBufferQueueItf queueItf);
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void RecorderSimpleBufferQueueCallbackHandler(
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SLAndroidSimpleBufferQueueItf queueItf);
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void CheckErr(SLresult res);
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@ -218,99 +239,79 @@ private:
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// Init
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WebRtc_Word32 InitSampleRate();
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// Threads
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static bool RecThreadFunc(void*);
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static bool PlayThreadFunc(void*);
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bool RecThreadProcess();
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bool PlayThreadProcess();
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// Misc
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AudioDeviceBuffer* _ptrAudioBuffer;
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CriticalSectionWrapper& _critSect;
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WebRtc_Word32 _id;
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AudioDeviceBuffer* voe_audio_buffer_;
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CriticalSectionWrapper& crit_sect_;
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WebRtc_Word32 id_;
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// audio unit
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SLObjectItf _slEngineObject;
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SLObjectItf sles_engine_;
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// playout device
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SLObjectItf _slPlayer;
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SLEngineItf _slEngine;
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SLPlayItf _slPlayerPlay;
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SLAndroidSimpleBufferQueueItf _slPlayerSimpleBufferQueue;
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SLObjectItf _slOutputMixObject;
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SLVolumeItf _slSpeakerVolume;
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SLObjectItf sles_player_;
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SLEngineItf sles_engine_itf_;
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SLPlayItf sles_player_itf_;
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SLAndroidSimpleBufferQueueItf sles_player_sbq_itf_;
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SLObjectItf sles_output_mixer_;
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SLVolumeItf sles_speaker_volume_;
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// recording device
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SLObjectItf _slRecorder;
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SLRecordItf _slRecorderRecord;
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SLAudioIODeviceCapabilitiesItf _slAudioIODeviceCapabilities;
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SLAndroidSimpleBufferQueueItf _slRecorderSimpleBufferQueue;
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SLDeviceVolumeItf _slMicVolume;
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SLObjectItf sles_recorder_;
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SLRecordItf sles_recorder_itf_;
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SLAndroidSimpleBufferQueueItf sles_recorder_sbq_itf_;
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SLDeviceVolumeItf sles_mic_volume_;
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WebRtc_UWord32 mic_dev_id_;
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WebRtc_UWord32 _micDeviceId;
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WebRtc_UWord32 _recQueueSeq;
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// Events
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EventWrapper& _timeEventRec;
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// Threads
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ThreadWrapper* _ptrThreadRec;
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WebRtc_UWord32 _recThreadID;
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// TODO(xians), remove the following flag
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bool _recThreadIsInitialized;
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// Playout buffer
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WebRtc_Word8 _playQueueBuffer[N_PLAY_QUEUE_BUFFERS][2
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* PLAY_BUF_SIZE_IN_SAMPLES];
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WebRtc_UWord32 _playQueueSeq;
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WebRtc_UWord32 play_warning_, play_error_;
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WebRtc_UWord32 rec_warning_, rec_error_;
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// States
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bool _recordingDeviceIsSpecified;
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bool _playoutDeviceIsSpecified;
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bool _initialized;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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bool _micIsInitialized;
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bool _speakerIsInitialized;
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// Warnings and errors
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WebRtc_UWord16 _playWarning;
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WebRtc_UWord16 _playError;
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WebRtc_UWord16 _recWarning;
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WebRtc_UWord16 _recError;
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bool is_recording_dev_specified_;
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bool is_playout_dev_specified_;
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bool is_initialized_;
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bool is_recording_;
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bool is_playing_;
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bool is_rec_initialized_;
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bool is_play_initialized_;
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bool is_mic_initialized_;
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bool is_speaker_initialized_;
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// Delay
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WebRtc_UWord16 _playoutDelay;
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WebRtc_UWord16 _recordingDelay;
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WebRtc_UWord16 playout_delay_;
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WebRtc_UWord16 recording_delay_;
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// AGC state
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bool _AGC;
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bool agc_enabled_;
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// The sampling rate to use with Audio Device Buffer
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WebRtc_UWord32 _adbSampleRate;
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// Stored device properties
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WebRtc_UWord32 _samplingRateIn; // Sampling frequency for Mic
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WebRtc_UWord32 _samplingRateOut; // Sampling frequency for Speaker
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WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value
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WebRtc_UWord32 _minSpeakerVolume; // The minimum speaker volume value
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bool _loudSpeakerOn;
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// Threads
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ThreadWrapper* rec_thread_;
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WebRtc_UWord32 rec_thread_id_;
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static bool RecThreadFunc(void* context);
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bool RecThreadFuncImpl();
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EventWrapper& rec_timer_;
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// Recording buffer used by the queues.
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int8_t rec_buffer_[N_REC_BUFFERS][2 * REC_BUF_SIZE_IN_SAMPLES];
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WebRtc_UWord32 mic_sampling_rate_;
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WebRtc_UWord32 speaker_sampling_rate_;
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WebRtc_UWord32 max_speaker_vol_;
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WebRtc_UWord32 min_speaker_vol_;
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bool loundspeaker_on_;
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// Queues accessed by both callback thread and recording thread after
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// recording has been started.
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std::queue<int8_t*> rec_worker_queue_;
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std::queue<int8_t*> rec_available_queue_;
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SLDataFormat_PCM player_pcm_;
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SLDataFormat_PCM record_pcm_;
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// Queue accssed by only callback thread after recording has been started.
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std::queue<int8_t*> rec_callback_queue_;
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std::queue<WebRtc_Word8*> rec_queue_;
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std::queue<WebRtc_Word8*> rec_voe_audio_queue_;
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std::queue<WebRtc_Word8*> rec_voe_ready_queue_;
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WebRtc_Word8 rec_buf_[N_REC_QUEUE_BUFFERS][
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N_REC_CHANNELS * sizeof(int16_t) * REC_BUF_SIZE_IN_SAMPLES];
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WebRtc_Word8 rec_voe_buf_[N_REC_QUEUE_BUFFERS][
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N_REC_CHANNELS * sizeof(int16_t) * REC_BUF_SIZE_IN_SAMPLES];
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// Flag to protect setting the recording thread priority multiple times.
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bool is_thread_priority_set_;
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std::queue<WebRtc_Word8*> play_queue_;
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WebRtc_Word8 play_buf_[N_PLAY_QUEUE_BUFFERS][
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N_PLAY_CHANNELS * sizeof(int16_t) * PLAY_BUF_SIZE_IN_SAMPLES];
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_OPENSLES_ANDROID_H
|
||||
#endif // SRC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_OPENSLES_ANDROID_H_
|
||||
|
||||
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