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Remove Soundclip handling from libjingle.

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BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}
This commit is contained in:
Fredrik Solenberg 2015-05-19 11:37:56 +02:00
parent 1ab67aef80
commit ccb49e79fd
17 changed files with 15 additions and 514 deletions
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2
talk/app/webrtc/peerconnectionfactory.cc
View File

@ -276,7 +276,7 @@ rtc::Thread* PeerConnectionFactory::worker_thread() {
cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
return cricket::WebRtcMediaEngineFactory::Create(
default_adm_.get(), NULL, video_encoder_factory_.get(),
default_adm_.get(), video_encoder_factory_.get(),
video_decoder_factory_.get());
}

2
talk/libjingle.gyp
View File

@ -673,8 +673,6 @@
'session/media/mediasink.h',
'session/media/rtcpmuxfilter.cc',
'session/media/rtcpmuxfilter.h',
'session/media/soundclip.cc',
'session/media/soundclip.h',
'session/media/srtpfilter.cc',
'session/media/srtpfilter.h',
'session/media/typingmonitor.cc',

6
talk/media/base/fakemediaengine.h
View File

@ -630,11 +630,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
int max_bps_;
};
class FakeSoundclipMedia : public SoundclipMedia {
public:
virtual bool PlaySound(const char* buf, int len, int flags) { return true; }
};
class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
public:
explicit FakeDataMediaChannel(void* unused)
@ -784,7 +779,6 @@ class FakeVoiceEngine : public FakeBaseEngine {
void UnregisterChannel(VoiceMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
SoundclipMedia* CreateSoundclip() { return new FakeSoundclipMedia(); }
const std::vector<AudioCodec>& codecs() { return codecs_; }
void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }

1
talk/media/base/filemediaengine.h
View File

@ -88,7 +88,6 @@ class FileMediaEngine : public MediaEngineInterface {
virtual VoiceMediaChannel* CreateChannel();
virtual VideoMediaChannel* CreateVideoChannel(const VideoOptions& options,
VoiceMediaChannel* voice_ch);
virtual SoundclipMedia* CreateSoundclip() { return NULL; }
virtual AudioOptions GetAudioOptions() const { return AudioOptions(); }
virtual bool SetAudioOptions(const AudioOptions& options) { return true; }
virtual bool SetAudioDelayOffset(int offset) { return true; }

1
talk/media/base/filemediaengine_unittest.cc
View File

@ -215,7 +215,6 @@ TEST_F(FileMediaEngineTest, TestDefaultImplementation) {
EXPECT_EQ(0, engine_->GetCapabilities());
EXPECT_TRUE(NULL == voice_channel_.get());
EXPECT_TRUE(NULL == video_channel_.get());
EXPECT_TRUE(NULL == engine_->CreateSoundclip());
cricket::AudioOptions audio_options;
EXPECT_TRUE(engine_->SetAudioOptions(audio_options));
VideoEncoderConfig video_encoder_config;

16
talk/media/base/mediachannel.h
View File

@ -446,22 +446,6 @@ struct VideoOptions {
Settable<int> screencast_min_bitrate;
};
// A class for playing out soundclips.
class SoundclipMedia {
public:
enum SoundclipFlags {
SF_LOOP = 1,
};
virtual ~SoundclipMedia() {}
// Plays a sound out to the speakers with the given audio stream. The stream
// must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
// on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
// Returns whether it was successful.
virtual bool PlaySound(const char *clip, int len, int flags) = 0;
};
struct RtpHeaderExtension {
RtpHeaderExtension() : id(0) {}
RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}

10
talk/media/base/mediaengine.h
View File

@ -84,9 +84,6 @@ class MediaEngineInterface {
const VideoOptions& options,
VoiceMediaChannel* voice_media_channel) = 0;
// Creates a soundclip object for playing sounds on. Returns NULL on failure.
virtual SoundclipMedia *CreateSoundclip() = 0;
// Configuration
// Gets global audio options.
virtual AudioOptions GetAudioOptions() const = 0;
@ -101,7 +98,6 @@ class MediaEngineInterface {
= 0;
// Device selection
// TODO(tschmelcher): Add method for selecting the soundclip device.
virtual bool SetSoundDevices(const Device* in_device,
const Device* out_device) = 0;
@ -193,9 +189,6 @@ class CompositeMediaEngine : public MediaEngineInterface {
VoiceMediaChannel* channel) {
return video_.CreateChannel(options, channel);
}
virtual SoundclipMedia *CreateSoundclip() {
return voice_.CreateSoundclip();
}
virtual AudioOptions GetAudioOptions() const {
return voice_.GetOptions();
@ -279,9 +272,6 @@ class NullVoiceEngine {
VoiceMediaChannel* CreateChannel() {
return NULL;
}
SoundclipMedia* CreateSoundclip() {
return NULL;
}
bool SetDelayOffset(int offset) { return true; }
AudioOptions GetOptions() const { return AudioOptions(); }
bool SetOptions(const AudioOptions& options) { return true; }

1
talk/media/other/linphonemediaengine.h
View File

@ -68,7 +68,6 @@ class LinphoneMediaEngine : public MediaEngineInterface {
virtual int GetCapabilities();
virtual VoiceMediaChannel* CreateChannel();
virtual VideoMediaChannel* CreateVideoChannel(VoiceMediaChannel* voice_ch);
virtual SoundclipMedia* CreateSoundclip() { return NULL; }
virtual bool SetAudioOptions(int options) { return true; }
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return true;

9
talk/media/webrtc/webrtcmediaengine.cc
View File

@ -37,10 +37,9 @@ class WebRtcMediaEngine2
: public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2> {
public:
WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory) {
voice_.SetAudioDeviceModule(adm, adm_sc);
voice_.SetAudioDeviceModule(adm);
video_.SetExternalDecoderFactory(decoder_factory);
video_.SetExternalEncoderFactory(encoder_factory);
}
@ -51,10 +50,9 @@ class WebRtcMediaEngine2
WRME_EXPORT
cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
cricket::WebRtcVideoEncoderFactory* encoder_factory,
cricket::WebRtcVideoDecoderFactory* decoder_factory) {
return new cricket::WebRtcMediaEngine2(adm, adm_sc, encoder_factory,
return new cricket::WebRtcMediaEngine2(adm, encoder_factory,
decoder_factory);
}
@ -69,10 +67,9 @@ namespace cricket {
// ChannelManager.
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory) {
return CreateWebRtcMediaEngine(adm, adm_sc, encoder_factory, decoder_factory);
return CreateWebRtcMediaEngine(adm, encoder_factory, decoder_factory);
}
} // namespace cricket

12
talk/media/webrtc/webrtcmediaengine.h
View File

@ -45,7 +45,7 @@ class WebRtcVideoEncoderFactory;
WRME_EXPORT
cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
webrtc::AudioDeviceModule* adm, webrtc::AudioDeviceModule* adm_sc,
webrtc::AudioDeviceModule* adm,
cricket::WebRtcVideoEncoderFactory* encoder_factory,
cricket::WebRtcVideoDecoderFactory* decoder_factory);
@ -69,7 +69,6 @@ class WebRtcMediaEngineFactory {
// !defined(LIBPEERCONNECTION_IMPLEMENTATION)
static MediaEngineInterface* Create(
webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory);
};
@ -88,11 +87,10 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
public:
DelegatingWebRtcMediaEngine(
webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory)
: delegate_(CreateWebRtcMediaEngine(
adm, adm_sc, encoder_factory, decoder_factory)) {
adm, encoder_factory, decoder_factory)) {
}
virtual ~DelegatingWebRtcMediaEngine() {
DestroyWebRtcMediaEngine(delegate_);
@ -110,9 +108,6 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
VoiceMediaChannel* voice_media_channel) override {
return delegate_->CreateVideoChannel(options, voice_media_channel);
}
SoundclipMedia* CreateSoundclip() override {
return delegate_->CreateSoundclip();
}
AudioOptions GetAudioOptions() const override {
return delegate_->GetAudioOptions();
}
@ -186,11 +181,10 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
// ChannelManager.
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc,
WebRtcVideoEncoderFactory* encoder_factory,
WebRtcVideoDecoderFactory* decoder_factory) {
return new cricket::DelegatingWebRtcMediaEngine(
adm, adm_sc, encoder_factory, decoder_factory);
adm, encoder_factory, decoder_factory);
}
} // namespace cricket

195
talk/media/webrtc/webrtcvoiceengine.cc
View File

@ -103,7 +103,6 @@ static const CodecPref kCodecPrefs[] = {
#ifdef WIN32
static const int kDefaultAudioDeviceId = -1;
static const int kDefaultSoundclipDeviceId = -2;
#else
static const int kDefaultAudioDeviceId = 0;
#endif
@ -368,103 +367,10 @@ static std::string GetEnableString(bool enable) {
return enable ? "enable" : "disable";
}
class WebRtcSoundclipMedia : public SoundclipMedia {
public:
explicit WebRtcSoundclipMedia(WebRtcVoiceEngine *engine)
: engine_(engine), webrtc_channel_(-1) {
engine_->RegisterSoundclip(this);
}
~WebRtcSoundclipMedia() override {
engine_->UnregisterSoundclip(this);
if (webrtc_channel_ != -1) {
// We shouldn't have to call Disable() here. DeleteChannel() should call
// StopPlayout() while deleting the channel. We should fix the bug
// inside WebRTC and remove the Disable() call bellow. This work is
// tracked by bug http://b/issue?id=5382855.
PlaySound(NULL, 0, 0);
Disable();
if (engine_->voe_sc()->base()->DeleteChannel(webrtc_channel_)
== -1) {
LOG_RTCERR1(DeleteChannel, webrtc_channel_);
}
}
}
bool Init() {
if (!engine_->voe_sc()) {
return false;
}
webrtc_channel_ = engine_->CreateSoundclipVoiceChannel();
if (webrtc_channel_ == -1) {
LOG_RTCERR0(CreateChannel);
return false;
}
return true;
}
bool Enable() {
if (engine_->voe_sc()->base()->StartPlayout(webrtc_channel_) == -1) {
LOG_RTCERR1(StartPlayout, webrtc_channel_);
return false;
}
return true;
}
bool Disable() {
if (engine_->voe_sc()->base()->StopPlayout(webrtc_channel_) == -1) {
LOG_RTCERR1(StopPlayout, webrtc_channel_);
return false;
}
return true;
}
bool PlaySound(const char* buf, int len, int flags) override {
// The voe file api is not available in chrome.
if (!engine_->voe_sc()->file()) {
return false;
}
// Must stop playing the current sound (if any), because we are about to
// modify the stream.
if (engine_->voe_sc()->file()->StopPlayingFileLocally(webrtc_channel_)
== -1) {
LOG_RTCERR1(StopPlayingFileLocally, webrtc_channel_);
return false;
}
if (buf) {
stream_.reset(new WebRtcSoundclipStream(buf, len));
stream_->set_loop((flags & SF_LOOP) != 0);
stream_->Rewind();
// Play it.
if (engine_->voe_sc()->file()->StartPlayingFileLocally(
webrtc_channel_, stream_.get()) == -1) {
LOG_RTCERR2(StartPlayingFileLocally, webrtc_channel_, stream_.get());
LOG(LS_ERROR) << "Unable to start soundclip";
return false;
}
} else {
stream_.reset();
}
return true;
}
int GetLastEngineError() const { return engine_->voe_sc()->error(); }
private:
WebRtcVoiceEngine *engine_;
int webrtc_channel_;
rtc::scoped_ptr<WebRtcSoundclipStream> stream_;
};
WebRtcVoiceEngine::WebRtcVoiceEngine()
: voe_wrapper_(new VoEWrapper()),
voe_wrapper_sc_(new VoEWrapper()),
voe_wrapper_sc_initialized_(false),
tracing_(new VoETraceWrapper()),
adm_(NULL),
adm_sc_(NULL),
log_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
@ -474,14 +380,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine()
}
WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
VoEWrapper* voe_wrapper_sc,
VoETraceWrapper* tracing)
: voe_wrapper_(voe_wrapper),
voe_wrapper_sc_(voe_wrapper_sc),
voe_wrapper_sc_initialized_(false),
tracing_(tracing),
adm_(NULL),
adm_sc_(NULL),
log_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
@ -593,11 +495,6 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
adm_->Release();
adm_ = NULL;
}
if (adm_sc_) {
voe_wrapper_sc_.reset();
adm_sc_->Release();
adm_sc_ = NULL;
}
// Test to see if the media processor was deregistered properly
DCHECK(SignalRxMediaFrame.is_empty());
@ -673,61 +570,12 @@ bool WebRtcVoiceEngine::InitInternal() {
return true;
}
bool WebRtcVoiceEngine::EnsureSoundclipEngineInit() {
if (voe_wrapper_sc_initialized_) {
return true;
}
// Note that, if initialization fails, voe_wrapper_sc_initialized_ will still
// be false, so subsequent calls to EnsureSoundclipEngineInit will
// probably just fail again. That's acceptable behavior.
#if defined(LINUX) && !defined(HAVE_LIBPULSE)
voe_wrapper_sc_->hw()->SetAudioDeviceLayer(webrtc::kAudioLinuxAlsa);
#endif
// Initialize the VoiceEngine instance that we'll use to play out sound clips.
if (voe_wrapper_sc_->base()->Init(adm_sc_) == -1) {
LOG_RTCERR0_EX(Init, voe_wrapper_sc_->error());
return false;
}
// On Windows, tell it to use the default sound (not communication) devices.
// First check whether there is a valid sound device for playback.
// TODO(juberti): Clean this up when we support setting the soundclip device.
#ifdef WIN32
// The SetPlayoutDevice may not be implemented in the case of external ADM.
// TODO(ronghuawu): We should only check the adm_sc_ here, but current
// PeerConnection interface never set the adm_sc_, so need to check both
// in order to determine if the external adm is used.
if (!adm_ && !adm_sc_) {
int num_of_devices = 0;
if (voe_wrapper_sc_->hw()->GetNumOfPlayoutDevices(num_of_devices) != -1 &&
num_of_devices > 0) {
if (voe_wrapper_sc_->hw()->SetPlayoutDevice(kDefaultSoundclipDeviceId)
== -1) {
LOG_RTCERR1_EX(SetPlayoutDevice, kDefaultSoundclipDeviceId,
voe_wrapper_sc_->error());
return false;
}
} else {
LOG(LS_WARNING) << "No valid sound playout device found.";
}
}
#endif
voe_wrapper_sc_initialized_ = true;
LOG(LS_INFO) << "Initialized WebRtc soundclip engine.";
return true;
}
void WebRtcVoiceEngine::Terminate() {
LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
initialized_ = false;
StopAecDump();
if (voe_wrapper_sc_) {
voe_wrapper_sc_initialized_ = false;
voe_wrapper_sc_->base()->Terminate();
}
voe_wrapper_->base()->Terminate();
desired_local_monitor_enable_ = false;
}
@ -745,20 +593,6 @@ VoiceMediaChannel *WebRtcVoiceEngine::CreateChannel() {
return ch;
}
SoundclipMedia *WebRtcVoiceEngine::CreateSoundclip() {
if (!EnsureSoundclipEngineInit()) {
LOG(LS_ERROR) << "Unable to create soundclip: soundclip engine failed to "
<< "initialize.";
return NULL;
}
WebRtcSoundclipMedia *soundclip = new WebRtcSoundclipMedia(this);
if (!soundclip->Init() || !soundclip->Enable()) {
delete soundclip;
return NULL;
}
return soundclip;
}
bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
if (!ApplyOptions(options)) {
return false;
@ -1532,19 +1366,6 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
}
}
void WebRtcVoiceEngine::RegisterSoundclip(WebRtcSoundclipMedia *soundclip) {
soundclips_.push_back(soundclip);
}
void WebRtcVoiceEngine::UnregisterSoundclip(WebRtcSoundclipMedia *soundclip) {
SoundclipList::iterator i = std::find(soundclips_.begin(),
soundclips_.end(),
soundclip);
if (i != soundclips_.end()) {
soundclips_.erase(i);
}
}
// Adjusts the default AGC target level by the specified delta.
// NB: If we start messing with other config fields, we'll want
// to save the current webrtc::AgcConfig as well.
@ -1563,8 +1384,7 @@ bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
return true;
}
bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc) {
bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
if (initialized_) {
LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
return false;
@ -1577,15 +1397,6 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
adm_ = adm;
adm_->AddRef();
}
if (adm_sc_) {
adm_sc_->Release();
adm_sc_ = NULL;
}
if (adm_sc) {
adm_sc_ = adm_sc;
adm_sc_->AddRef();
}
return true;
}
@ -1791,10 +1602,6 @@ int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
return CreateVoiceChannel(voe_wrapper_.get());
}
int WebRtcVoiceEngine::CreateSoundclipVoiceChannel() {
return CreateVoiceChannel(voe_wrapper_sc_.get());
}
class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
: public AudioRenderer::Sink {
public:

25
talk/media/webrtc/webrtcvoiceengine.h
View File

@ -91,7 +91,6 @@ class AudioRenderer;
class VoETraceWrapper;
class VoEWrapper;
class VoiceProcessor;
class WebRtcSoundclipMedia;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
@ -103,9 +102,7 @@ class WebRtcVoiceEngine
public:
WebRtcVoiceEngine();
// Dependency injection for testing.
WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
VoEWrapper* voe_wrapper_sc,
VoETraceWrapper* tracing);
WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
~WebRtcVoiceEngine();
bool Init(rtc::Thread* worker_thread);
void Terminate();
@ -113,8 +110,6 @@ class WebRtcVoiceEngine
int GetCapabilities();
VoiceMediaChannel* CreateChannel();
SoundclipMedia* CreateSoundclip();
AudioOptions GetOptions() const { return options_; }
bool SetOptions(const AudioOptions& options);
// Overrides, when set, take precedence over the options on a
@ -166,21 +161,15 @@ class WebRtcVoiceEngine
void RegisterChannel(WebRtcVoiceMediaChannel *channel);
void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
// May only be called by WebRtcSoundclipMedia.
void RegisterSoundclip(WebRtcSoundclipMedia *channel);
void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
// Called by WebRtcVoiceMediaChannel to set a gain offset from
// the default AGC target level.
bool AdjustAgcLevel(int delta);
VoEWrapper* voe() { return voe_wrapper_.get(); }
VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
int GetLastEngineError();
// Set the external ADMs. This can only be called before Init.
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc);
// Set the external ADM. This can only be called before Init.
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
// Starts AEC dump using existing file.
bool StartAecDump(rtc::PlatformFile file);
@ -190,10 +179,8 @@ class WebRtcVoiceEngine
// Create a VoiceEngine Channel.
int CreateMediaVoiceChannel();
int CreateSoundclipVoiceChannel();
private:
typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
typedef sigslot::
signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
@ -202,7 +189,6 @@ class WebRtcVoiceEngine
void ConstructCodecs();
bool GetVoeCodec(int index, webrtc::CodecInst* codec);
bool InitInternal();
bool EnsureSoundclipEngineInit();
void SetTraceFilter(int filter);
void SetTraceOptions(const std::string& options);
// Applies either options or overrides. Every option that is "set"
@ -250,13 +236,9 @@ class WebRtcVoiceEngine
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
// A secondary instance, for playing out soundclips (on the 'ring' device).
rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
bool voe_wrapper_sc_initialized_;
rtc::scoped_ptr<VoETraceWrapper> tracing_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_;
webrtc::AudioDeviceModule* adm_sc_;
int log_filter_;
std::string log_options_;
bool is_dumping_aec_;
@ -264,7 +246,6 @@ class WebRtcVoiceEngine
std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool desired_local_monitor_enable_;
rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
SoundclipList soundclips_;
ChannelList channels_;
// channels_ can be read from WebRtc callback thread. We need a lock on that
// callback as well as the RegisterChannel/UnregisterChannel.

40
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -128,12 +128,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
WebRtcVoiceEngineTestFake()
: voe_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
voe_sc_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
trace_wrapper_(new FakeVoETraceWrapper()),
engine_(new FakeVoEWrapper(&voe_),
new FakeVoEWrapper(&voe_sc_),
trace_wrapper_),
channel_(NULL), soundclip_(NULL) {
channel_(NULL) {
options_conference_.conference_mode.Set(true);
options_adjust_agc_.adjust_agc_delta.Set(-10);
}
@ -168,7 +166,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
channel_->OnPacketReceived(&packet, rtc::PacketTime());
}
void TearDown() override {
delete soundclip_;
delete channel_;
engine_.Terminate();
}
@ -335,11 +332,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
protected:
cricket::FakeWebRtcVoiceEngine voe_;
cricket::FakeWebRtcVoiceEngine voe_sc_;
FakeVoETraceWrapper* trace_wrapper_;
cricket::WebRtcVoiceEngine engine_;
cricket::VoiceMediaChannel* channel_;
cricket::SoundclipMedia* soundclip_;
cricket::AudioOptions options_conference_;
cricket::AudioOptions options_adjust_agc_;
@ -348,14 +343,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
// Tests that our stub library "works".
TEST_F(WebRtcVoiceEngineTestFake, StartupShutdown) {
EXPECT_FALSE(voe_.IsInited());
EXPECT_FALSE(voe_sc_.IsInited());
EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
EXPECT_TRUE(voe_.IsInited());
// The soundclip engine is lazily initialized.
EXPECT_FALSE(voe_sc_.IsInited());
engine_.Terminate();
EXPECT_FALSE(voe_.IsInited());
EXPECT_FALSE(voe_sc_.IsInited());
}
// Tests that we can create and destroy a channel.
@ -2652,35 +2643,6 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayRingbackWithMultipleStreams) {
EXPECT_EQ(0, voe_.IsPlayingFileLocally(channel_num));
}
// Tests creating soundclips, and make sure they come from the right engine.
TEST_F(WebRtcVoiceEngineTestFake, CreateSoundclip) {
EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
EXPECT_FALSE(voe_sc_.IsInited());
soundclip_ = engine_.CreateSoundclip();
EXPECT_TRUE(voe_sc_.IsInited());
ASSERT_TRUE(soundclip_ != NULL);
EXPECT_EQ(0, voe_.GetNumChannels());
EXPECT_EQ(1, voe_sc_.GetNumChannels());
int channel_num = voe_sc_.GetLastChannel();
EXPECT_TRUE(voe_sc_.GetPlayout(channel_num));
delete soundclip_;
soundclip_ = NULL;
EXPECT_EQ(0, voe_sc_.GetNumChannels());
// Make sure the soundclip engine is uninitialized on shutdown, now that
// we've initialized it by creating a soundclip.
engine_.Terminate();
EXPECT_FALSE(voe_sc_.IsInited());
}
// Tests playing out a fake sound.
TEST_F(WebRtcVoiceEngineTestFake, PlaySoundclip) {
static const char kZeroes[16000] = {};
EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
soundclip_ = engine_.CreateSoundclip();
ASSERT_TRUE(soundclip_ != NULL);
EXPECT_TRUE(soundclip_->PlaySound(kZeroes, sizeof(kZeroes), 0));
}
TEST_F(WebRtcVoiceEngineTestFake, MediaEngineCallbackOnError) {
rtc::scoped_ptr<ChannelErrorListener> listener;
cricket::WebRtcVoiceMediaChannel* media_channel;

43
talk/session/media/channelmanager.cc
View File

@ -41,7 +41,6 @@
#ifdef HAVE_SCTP
#include "talk/media/sctp/sctpdataengine.h"
#endif
#include "talk/session/media/soundclip.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
@ -311,9 +310,6 @@ void ChannelManager::Terminate_w() {
while (!voice_channels_.empty()) {
DestroyVoiceChannel_w(voice_channels_.back(), nullptr);
}
while (!soundclips_.empty()) {
DestroySoundclip_w(soundclips_.back());
}
if (!SetCaptureDevice_w(NULL)) {
LOG(LS_WARNING) << "failed to delete video capturer";
}
@ -504,45 +500,6 @@ void ChannelManager::DestroyDataChannel_w(DataChannel* data_channel) {
delete data_channel;
}
Soundclip* ChannelManager::CreateSoundclip() {
return worker_thread_->Invoke<Soundclip*>(
Bind(&ChannelManager::CreateSoundclip_w, this));
}
Soundclip* ChannelManager::CreateSoundclip_w() {
ASSERT(initialized_);
ASSERT(worker_thread_ == rtc::Thread::Current());
SoundclipMedia* soundclip_media = media_engine_->CreateSoundclip();
if (!soundclip_media) {
return NULL;
}
Soundclip* soundclip = new Soundclip(worker_thread_, soundclip_media);
soundclips_.push_back(soundclip);
return soundclip;
}
void ChannelManager::DestroySoundclip(Soundclip* soundclip) {
if (soundclip) {
worker_thread_->Invoke<void>(
Bind(&ChannelManager::DestroySoundclip_w, this, soundclip));
}
}
void ChannelManager::DestroySoundclip_w(Soundclip* soundclip) {
// Destroy soundclip.
ASSERT(initialized_);
Soundclips::iterator it = std::find(soundclips_.begin(),
soundclips_.end(), soundclip);
ASSERT(it != soundclips_.end());
if (it == soundclips_.end())
return;
soundclips_.erase(it);
delete soundclip;
}
bool ChannelManager::GetAudioOptions(std::string* in_name,
std::string* out_name,
AudioOptions* options) {

13
talk/session/media/channelmanager.h
View File

@ -44,7 +44,6 @@ namespace cricket {
const int kDefaultAudioDelayOffset = 0;
class Soundclip;
class VideoProcessor;
class VoiceChannel;
class VoiceProcessor;
@ -129,15 +128,9 @@ class ChannelManager : public rtc::MessageHandler,
// Destroys a data channel created with the Create API.
void DestroyDataChannel(DataChannel* data_channel);
// Creates a soundclip.
Soundclip* CreateSoundclip();
// Destroys a soundclip created with the Create API.
void DestroySoundclip(Soundclip* soundclip);
// Indicates whether any channels exist.
bool has_channels() const {
return (!voice_channels_.empty() || !video_channels_.empty() ||
!soundclips_.empty());
return (!voice_channels_.empty() || !video_channels_.empty());
}
// Configures the audio and video devices. A null pointer can be passed to
@ -253,7 +246,6 @@ class ChannelManager : public rtc::MessageHandler,
typedef std::vector<VoiceChannel*> VoiceChannels;
typedef std::vector<VideoChannel*> VideoChannels;
typedef std::vector<DataChannel*> DataChannels;
typedef std::vector<Soundclip*> Soundclips;
void Construct(MediaEngineInterface* me,
DataEngineInterface* dme,
@ -277,8 +269,6 @@ class ChannelManager : public rtc::MessageHandler,
BaseSession* session, const std::string& content_name,
bool rtcp, DataChannelType data_channel_type);
void DestroyDataChannel_w(DataChannel* data_channel);
Soundclip* CreateSoundclip_w();
void DestroySoundclip_w(Soundclip* soundclip);
bool SetAudioOptions_w(const AudioOptions& options, int delay_offset,
const Device* in_dev, const Device* out_dev);
bool SetEngineAudioOptions_w(const AudioOptions& options);
@ -306,7 +296,6 @@ class ChannelManager : public rtc::MessageHandler,
VoiceChannels voice_channels_;
VideoChannels video_channels_;
DataChannels data_channels_;
Soundclips soundclips_;
std::string audio_in_device_;
std::string audio_out_device_;

83
talk/session/media/soundclip.cc
View File

@ -1,82 +1 @@
/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/soundclip.h"
namespace cricket {
enum {
MSG_PLAYSOUND = 1,
};
struct PlaySoundMessageData : rtc::MessageData {
PlaySoundMessageData(const void *c,
int l,
SoundclipMedia::SoundclipFlags f)
: clip(c),
len(l),
flags(f),
result(false) {
}
const void *clip;
int len;
SoundclipMedia::SoundclipFlags flags;
bool result;
};
Soundclip::Soundclip(rtc::Thread *thread, SoundclipMedia *soundclip_media)
: worker_thread_(thread),
soundclip_media_(soundclip_media) {
}
bool Soundclip::PlaySound(const void *clip,
int len,
SoundclipMedia::SoundclipFlags flags) {
PlaySoundMessageData data(clip, len, flags);
worker_thread_->Send(this, MSG_PLAYSOUND, &data);
return data.result;
}
bool Soundclip::PlaySound_w(const void *clip,
int len,
SoundclipMedia::SoundclipFlags flags) {
return soundclip_media_->PlaySound(static_cast<const char *>(clip),
len,
flags);
}
void Soundclip::OnMessage(rtc::Message *message) {
ASSERT(message->message_id == MSG_PLAYSOUND);
PlaySoundMessageData *data =
static_cast<PlaySoundMessageData *>(message->pdata);
data->result = PlaySound_w(data->clip,
data->len,
data->flags);
}
} // namespace cricket
// TODO(solenberg): Remove this file when it's no longer built in Chromium.

70
talk/session/media/soundclip.h
View File

@ -1,70 +1,2 @@
/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
// TODO(solenberg): Remove this file when it's no longer built in Chromium.
#ifndef TALK_SESSION_MEDIA_SOUNDCLIP_H_
#define TALK_SESSION_MEDIA_SOUNDCLIP_H_
#include "talk/media/base/mediaengine.h"
#include "webrtc/base/scoped_ptr.h"
namespace rtc {
class Thread;
}
namespace cricket {
// Soundclip wraps SoundclipMedia to support marshalling calls to the proper
// thread.
class Soundclip : private rtc::MessageHandler {
public:
Soundclip(rtc::Thread* thread, SoundclipMedia* soundclip_media);
// Plays a sound out to the speakers with the given audio stream. The stream
// must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
// on this Soundclip, it is stopped. If clip is NULL, nothing is played.
// Returns whether it was successful.
bool PlaySound(const void* clip,
int len,
SoundclipMedia::SoundclipFlags flags);
private:
bool PlaySound_w(const void* clip,
int len,
SoundclipMedia::SoundclipFlags flags);
// From MessageHandler
virtual void OnMessage(rtc::Message* message);
rtc::Thread* worker_thread_;
rtc::scoped_ptr<SoundclipMedia> soundclip_media_;
};
} // namespace cricket
#endif // TALK_SESSION_MEDIA_SOUNDCLIP_H_