After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.

CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tkchin@webrtc.org
2014-07-15 20:20:47 +00:00
parent 47218956fc
commit 122caa51b1
8 changed files with 491 additions and 438 deletions

View File

@ -98,7 +98,7 @@
'linux/audio_mixer_manager_alsa_linux.h',
'linux/latebindingsymboltable_linux.cc',
'linux/latebindingsymboltable_linux.h',
'ios/audio_device_ios.cc',
'ios/audio_device_ios.mm',
'ios/audio_device_ios.h',
'ios/audio_device_utility_ios.cc',
'ios/audio_device_utility_ios.h',
@ -175,7 +175,7 @@
}],
],
}],
['OS=="mac" or OS=="ios"', {
['OS=="mac"', {
'link_settings': {
'libraries': [
'$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
@ -183,6 +183,19 @@
],
},
}],
['OS=="ios"', {
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AudioToolbox',
'-framework AVFoundation',
],
},
},
}],
['OS=="win"', {
'link_settings': {
'libraries': [

View File

@ -349,15 +349,15 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
#if defined(WEBRTC_IOS)
if (audioLayer == kPlatformDefaultAudio)
{
// Create *iPhone Audio* implementation
ptrAudioDevice = new AudioDeviceIPhone(Id());
// Create iOS Audio Device implementation.
ptrAudioDevice = new AudioDeviceIOS(Id());
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "iPhone Audio APIs will be utilized");
}
if (ptrAudioDevice != NULL)
{
// Create the Mac implementation of the Device Utility.
ptrAudioDeviceUtility = new AudioDeviceUtilityIPhone(Id());
// Create iOS Device Utility implementation.
ptrAudioDeviceUtility = new AudioDeviceUtilityIOS(Id());
}
// END #if defined(WEBRTC_IOS)

View File

@ -18,7 +18,7 @@ namespace webrtc
class AudioDeviceUtility
{
public:
public:
static uint32_t GetTimeInMS();
static void WaitForKey();
static bool StringCompare(const char* str1,

View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H
#include <AudioUnit/AudioUnit.h>
@ -32,14 +32,14 @@ const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
// Number of 10 ms recording blocks in recording buffer
const uint16_t N_REC_BUFFERS = 20;
class AudioDeviceIPhone : public AudioDeviceGeneric {
public:
AudioDeviceIPhone(const int32_t id);
~AudioDeviceIPhone();
class AudioDeviceIOS : public AudioDeviceGeneric {
public:
AudioDeviceIOS(const int32_t id);
~AudioDeviceIOS();
// Retrieve the currently utilized audio layer
virtual int32_t
ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
virtual int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
virtual int32_t Init();
@ -58,8 +58,7 @@ public:
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index);
virtual int32_t
SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
virtual int32_t SetRecordingDevice(uint16_t index);
virtual int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device);
@ -109,8 +108,7 @@ public:
virtual int32_t MicrophoneVolume(uint32_t& volume) const;
virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
virtual int32_t
MicrophoneVolumeStepSize(uint16_t& stepSize) const;
virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool& available);
@ -136,8 +134,7 @@ public:
virtual int32_t StereoRecording(bool& enabled) const;
// Delay information and control
virtual int32_t
SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
uint16_t sizeMS);
virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
uint16_t& sizeMS) const;
@ -147,7 +144,7 @@ public:
// CPU load
virtual int32_t CPULoad(uint16_t& load) const;
public:
public:
virtual bool PlayoutWarning() const;
virtual bool PlayoutError() const;
virtual bool RecordingWarning() const;
@ -157,17 +154,17 @@ public:
virtual void ClearRecordingWarning();
virtual void ClearRecordingError();
public:
public:
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
// Reset Audio Deivce (for mobile devices only)
// Reset Audio Device (for mobile devices only)
virtual int32_t ResetAudioDevice();
// enable or disable loud speaker (for iphone only)
virtual int32_t SetLoudspeakerStatus(bool enable);
virtual int32_t GetLoudspeakerStatus(bool& enabled) const;
private:
private:
void Lock() {
_critSect.Enter();
}
@ -212,7 +209,7 @@ private:
static bool RunCapture(void* ptrThis);
bool CaptureWorkerThread();
private:
private:
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
@ -223,8 +220,9 @@ private:
int32_t _id;
AudioUnit _auVoiceProcessing;
void* _audioInterruptionObserver;
private:
private:
bool _initialized;
bool _isShutDown;
bool _recording;
@ -260,8 +258,7 @@ private:
uint32_t _playoutBufferUsed; // How much is filled
// Recording buffers
int16_t
_recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES];
int16_t _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES];
uint32_t _recordingLength[N_REC_BUFFERS];
uint32_t _recordingSeqNumber[N_REC_BUFFERS];
uint32_t _recordingCurrentSeq;
@ -272,4 +269,4 @@ private:
} // namespace webrtc
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_IPHONE_H_
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IOS_H

View File

@ -15,7 +15,7 @@
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
AudioDeviceUtilityIPhone::AudioDeviceUtilityIPhone(const int32_t id)
AudioDeviceUtilityIOS::AudioDeviceUtilityIOS(const int32_t id)
:
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_id(id),
@ -24,7 +24,7 @@ AudioDeviceUtilityIPhone::AudioDeviceUtilityIPhone(const int32_t id)
"%s created", __FUNCTION__);
}
AudioDeviceUtilityIPhone::~AudioDeviceUtilityIPhone() {
AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS() {
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
"%s destroyed", __FUNCTION__);
CriticalSectionScoped lock(&_critSect);
@ -32,7 +32,7 @@ AudioDeviceUtilityIPhone::~AudioDeviceUtilityIPhone() {
delete &_critSect;
}
int32_t AudioDeviceUtilityIPhone::Init() {
int32_t AudioDeviceUtilityIOS::Init() {
WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
"%s", __FUNCTION__);

View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H
#include "webrtc/modules/audio_device/audio_device_utility.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
@ -17,15 +17,15 @@
namespace webrtc {
class CriticalSectionWrapper;
class AudioDeviceUtilityIPhone: public AudioDeviceUtility {
public:
AudioDeviceUtilityIPhone(const int32_t id);
AudioDeviceUtilityIPhone();
virtual ~AudioDeviceUtilityIPhone();
class AudioDeviceUtilityIOS: public AudioDeviceUtility {
public:
AudioDeviceUtilityIOS(const int32_t id);
AudioDeviceUtilityIOS();
virtual ~AudioDeviceUtilityIOS();
virtual int32_t Init();
private:
private:
CriticalSectionWrapper& _critSect;
int32_t _id;
AudioDeviceModule::ErrorCode _lastError;
@ -33,4 +33,4 @@ private:
} // namespace webrtc
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IPHONE_H
#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_IOS_H

View File

@ -43,6 +43,7 @@ using namespace webrtc::videocapturemodule;
_owner = owner;
_captureId = captureId;
_captureSession = [[AVCaptureSession alloc] init];
_captureSession.usesApplicationAudioSession = NO;
_captureChanging = NO;
_captureChangingCondition = [[NSCondition alloc] init];