Refactor handling of AudioOptions.
- Remove MediaEngineInterface::GetAudioOptions(), SetAudioOptions() and SetSoundDevices().
- Remove the WebRtcVoiceEngine infrastructure for those calls.
BUG=webrtc:4690
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1500633002
Cr-Commit-Position: refs/heads/master@{#10938}
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index ab0c5eb..b1f09aa 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -722,14 +722,6 @@
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
return rtc::scoped_refptr<webrtc::AudioState>();
}
- AudioOptions GetOptions() const {
- return options_;
- }
- bool SetOptions(const AudioOptions& options) {
- options_ = options;
- options_changed_ = true;
- return true;
- }
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const AudioOptions& options) {
@@ -751,21 +743,12 @@
const std::vector<AudioCodec>& codecs() { return codecs_; }
void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
- bool SetDevices(const Device* in_device, const Device* out_device) {
- in_device_ = (in_device) ? in_device->name : "";
- out_device_ = (out_device) ? out_device->name : "";
- options_changed_ = true;
- return true;
- }
-
bool GetOutputVolume(int* level) {
*level = output_volume_;
return true;
}
-
bool SetOutputVolume(int level) {
output_volume_ = level;
- options_changed_ = true;
return true;
}
@@ -783,9 +766,6 @@
std::vector<FakeVoiceMediaChannel*> channels_;
std::vector<AudioCodec> codecs_;
int output_volume_;
- std::string in_device_;
- std::string out_device_;
- AudioOptions options_;
friend class FakeMediaEngine;
};
@@ -863,10 +843,7 @@
class FakeMediaEngine :
public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
public:
- FakeMediaEngine() {
- voice_ = FakeVoiceEngine();
- video_ = FakeVideoEngine();
- }
+ FakeMediaEngine() {}
virtual ~FakeMediaEngine() {}
void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
@@ -892,20 +869,16 @@
return video_.GetChannel(index);
}
- AudioOptions audio_options() const { return voice_.options_; }
int output_volume() const { return voice_.output_volume_; }
const VideoEncoderConfig& default_video_encoder_config() const {
return video_.default_encoder_config_;
}
- const std::string& audio_in_device() const { return voice_.in_device_; }
- const std::string& audio_out_device() const { return voice_.out_device_; }
bool capture() const { return video_.capture_; }
bool options_changed() const {
- return voice_.options_changed_ || video_.options_changed_;
+ return video_.options_changed_;
}
void clear_options_changed() {
video_.options_changed_ = false;
- voice_.options_changed_ = false;
}
void set_fail_create_channel(bool fail) {
voice_.set_fail_create_channel(fail);
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 2509cb0..92d0fcb 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -553,7 +553,6 @@
enum SendFlags {
SEND_NOTHING,
- SEND_RINGBACKTONE,
SEND_MICROPHONE
};
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index c31fd72..cc033a7 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -88,20 +88,11 @@
webrtc::Call* call,
const VideoOptions& options) = 0;
- // Configuration
- // Gets global audio options.
- virtual AudioOptions GetAudioOptions() const = 0;
- // Sets global audio options. "options" are from AudioOptions, above.
- virtual bool SetAudioOptions(const AudioOptions& options) = 0;
// Sets the default (maximum) codec/resolution and encoder option to capture
// and encode video.
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config)
= 0;
- // Device selection
- virtual bool SetSoundDevices(const Device* in_device,
- const Device* out_device) = 0;
-
// Device configuration
// Gets the current speaker volume, as a value between 0 and 255.
virtual bool GetOutputVolume(int* level) = 0;
@@ -176,21 +167,10 @@
return video_.CreateChannel(call, options);
}
- virtual AudioOptions GetAudioOptions() const {
- return voice_.GetOptions();
- }
- virtual bool SetAudioOptions(const AudioOptions& options) {
- return voice_.SetOptions(options);
- }
virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
return video_.SetDefaultEncoderConfig(config);
}
- virtual bool SetSoundDevices(const Device* in_device,
- const Device* out_device) {
- return voice_.SetDevices(in_device, out_device);
- }
-
virtual bool GetOutputVolume(int* level) {
return voice_.GetOutputVolume(level);
}
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 65c3deb..b7a7262 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -46,15 +46,6 @@
namespace cricket {
-static const char kFakeDefaultDeviceName[] = "Fake Default";
-static const int kFakeDefaultDeviceId = -1;
-static const char kFakeDeviceName[] = "Fake Device";
-#ifdef WIN32
-static const int kFakeDeviceId = 0;
-#else
-static const int kFakeDeviceId = 1;
-#endif
-
static const int kOpusBandwidthNb = 4000;
static const int kOpusBandwidthMb = 6000;
static const int kOpusBandwidthWb = 8000;
@@ -538,18 +529,10 @@
}
// webrtc::VoEHardware
- WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
- return GetNumDevices(num);
- }
- WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) {
- return GetNumDevices(num);
- }
- WEBRTC_FUNC(GetRecordingDeviceName, (int i, char* name, char* guid)) {
- return GetDeviceName(i, name, guid);
- }
- WEBRTC_FUNC(GetPlayoutDeviceName, (int i, char* name, char* guid)) {
- return GetDeviceName(i, name, guid);
- }
+ WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
+ WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
+ WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
+ WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
WEBRTC_STUB(SetPlayoutDevice, (int));
WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
@@ -808,41 +791,6 @@
}
private:
- int GetNumDevices(int& num) {
-#ifdef WIN32
- num = 1;
-#else
- // On non-Windows platforms VE adds a special entry for the default device,
- // so if there is one physical device then there are two entries in the
- // list.
- num = 2;
-#endif
- return 0;
- }
-
- int GetDeviceName(int i, char* name, char* guid) {
- const char *s;
-#ifdef WIN32
- if (0 == i) {
- s = kFakeDeviceName;
- } else {
- return -1;
- }
-#else
- // See comment above.
- if (0 == i) {
- s = kFakeDefaultDeviceName;
- } else if (1 == i) {
- s = kFakeDeviceName;
- } else {
- return -1;
- }
-#endif
- strcpy(name, s);
- guid[0] = '\0';
- return 0;
- }
-
bool inited_;
int last_channel_;
std::map<int, Channel*> channels_;
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index fd0fc4b..0b5bed1 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -68,21 +68,12 @@
const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
webrtc::kTraceInfo;
-// For Linux/Mac, using the default device is done by specifying index 0 for
-// VoE 4.0 and not -1 (which was the case for VoE 3.5).
-//
// On Windows Vista and newer, Microsoft introduced the concept of "Default
// Communications Device". This means that there are two types of default
// devices (old Wave Audio style default and Default Communications Device).
//
// On Windows systems which only support Wave Audio style default, uses either
// -1 or 0 to select the default device.
-//
-// On Windows systems which support both "Default Communication Device" and
-// old Wave Audio style default, use -1 for Default Communications Device and
-// -2 for Wave Audio style default, which is what we want to use for clips.
-// It's not clear yet whether the -2 index is handled properly on other OSes.
-
#ifdef WIN32
const int kDefaultAudioDeviceId = -1;
#else
@@ -275,28 +266,6 @@
voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
-// Gets the default set of options applied to the engine. Historically, these
-// were supplied as a combination of flags from the channel manager (ec, agc,
-// ns, and highpass) and the rest hardcoded in InitInternal.
-AudioOptions GetDefaultEngineOptions() {
- AudioOptions options;
- options.echo_cancellation = rtc::Optional<bool>(true);
- options.auto_gain_control = rtc::Optional<bool>(true);
- options.noise_suppression = rtc::Optional<bool>(true);
- options.highpass_filter = rtc::Optional<bool>(true);
- options.stereo_swapping = rtc::Optional<bool>(false);
- options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
- options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
- options.typing_detection = rtc::Optional<bool>(true);
- options.adjust_agc_delta = rtc::Optional<int>(0);
- options.experimental_agc = rtc::Optional<bool>(false);
- options.extended_filter_aec = rtc::Optional<bool>(false);
- options.delay_agnostic_aec = rtc::Optional<bool>(false);
- options.experimental_ns = rtc::Optional<bool>(false);
- options.aec_dump = rtc::Optional<bool>(false);
- return options;
-}
-
webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
webrtc::AudioState::Config config;
config.voice_engine = voe_wrapper->engine();
@@ -510,15 +479,13 @@
signal_thread_checker_.DetachFromThread();
std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
+ voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
webrtc::Trace::SetTraceCallback(this);
// Load our audio codec list.
codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
-
- options_ = GetDefaultEngineOptions();
- voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
}
WebRtcVoiceEngine::~WebRtcVoiceEngine() {
@@ -558,25 +525,20 @@
webrtc::Trace::set_level_filter(kDefaultTraceFilter);
// Save the default AGC configuration settings. This must happen before
- // calling SetOptions or the default will be overwritten.
+ // calling ApplyOptions or the default will be overwritten.
if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
LOG_RTCERR0(GetAgcConfig);
return false;
}
- // Set defaults for options, so that ApplyOptions applies them explicitly
- // when we clear option (channel) overrides. External clients can still
- // modify the defaults via SetOptions (on the media engine).
- if (!SetOptions(GetDefaultEngineOptions())) {
- return false;
- }
-
// Print our codec list again for the call diagnostic log
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
for (const AudioCodec& codec : codecs_) {
LOG(LS_INFO) << ToString(codec);
}
+ SetDefaultDevices();
+
initialized_ = true;
return true;
}
@@ -603,21 +565,30 @@
return new WebRtcVoiceMediaChannel(this, options, call);
}
-bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- if (!ApplyOptions(options)) {
- return false;
- }
- options_ = options;
- return true;
-}
-
-// AudioOptions defaults are set in InitInternal (for options with corresponding
-// MediaEngineInterface flags) and in SetOptions(int) for flagless options.
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
- AudioOptions options = options_in; // The options are modified below.
+
+ // Default engine options.
+ AudioOptions options;
+ options.echo_cancellation = rtc::Optional<bool>(true);
+ options.auto_gain_control = rtc::Optional<bool>(true);
+ options.noise_suppression = rtc::Optional<bool>(true);
+ options.highpass_filter = rtc::Optional<bool>(true);
+ options.stereo_swapping = rtc::Optional<bool>(false);
+ options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
+ options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
+ options.typing_detection = rtc::Optional<bool>(true);
+ options.adjust_agc_delta = rtc::Optional<int>(0);
+ options.experimental_agc = rtc::Optional<bool>(false);
+ options.extended_filter_aec = rtc::Optional<bool>(false);
+ options.delay_agnostic_aec = rtc::Optional<bool>(false);
+ options.experimental_ns = rtc::Optional<bool>(false);
+ options.aec_dump = rtc::Optional<bool>(false);
+
+ // Apply any given options on top.
+ options.SetAll(options_in);
+
// kEcConference is AEC with high suppression.
webrtc::EcModes ec_mode = webrtc::kEcConference;
webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
@@ -887,149 +858,36 @@
return true;
}
-// TODO(juberti): Refactor this so that the core logic can be used to set the
-// soundclip device. At that time, reinstate the soundclip pause/resume code.
-bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
- const Device* out_device) {
+void WebRtcVoiceEngine::SetDefaultDevices() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
#if !defined(IOS)
- int in_id = in_device ? rtc::FromString<int>(in_device->id) :
- kDefaultAudioDeviceId;
- int out_id = out_device ? rtc::FromString<int>(out_device->id) :
- kDefaultAudioDeviceId;
- // The device manager uses -1 as the default device, which was the case for
- // VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
-#ifndef WIN32
- if (-1 == in_id) {
- in_id = kDefaultAudioDeviceId;
- }
- if (-1 == out_id) {
- out_id = kDefaultAudioDeviceId;
- }
-#endif
+ int in_id = kDefaultAudioDeviceId;
+ int out_id = kDefaultAudioDeviceId;
+ LOG(LS_INFO) << "Setting microphone to (id=" << in_id
+ << ") and speaker to (id=" << out_id << ")";
- std::string in_name = (in_id != kDefaultAudioDeviceId) ?
- in_device->name : "Default device";
- std::string out_name = (out_id != kDefaultAudioDeviceId) ?
- out_device->name : "Default device";
- LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name
- << ") and speaker to (id=" << out_id << ", name=" << out_name
- << ")";
-
- // Must also pause all audio playback and capture.
bool ret = true;
- for (WebRtcVoiceMediaChannel* channel : channels_) {
- if (!channel->PausePlayout()) {
- LOG(LS_WARNING) << "Failed to pause playout";
- ret = false;
- }
- if (!channel->PauseSend()) {
- LOG(LS_WARNING) << "Failed to pause send";
- ret = false;
- }
- }
-
- // Find the recording device id in VoiceEngine and set recording device.
- if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) {
+ if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
+ LOG_RTCERR1(SetRecordingDevice, in_id);
ret = false;
}
- if (ret) {
- if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
- LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
- ret = false;
- }
- webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
- if (ap)
- ap->Initialize();
+ webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
+ if (ap) {
+ ap->Initialize();
}
- // Find the playout device id in VoiceEngine and set playout device.
- if (!FindWebRtcAudioDeviceId(false, out_name, out_id, &out_id)) {
- LOG(LS_WARNING) << "Failed to find VoiceEngine device id for " << out_name;
+ if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
+ LOG_RTCERR1(SetPlayoutDevice, out_id);
ret = false;
}
- if (ret) {
- if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
- LOG_RTCERR2(SetPlayoutDevice, out_name, out_id);
- ret = false;
- }
- }
-
- // Resume all audio playback and capture.
- for (WebRtcVoiceMediaChannel* channel : channels_) {
- if (!channel->ResumePlayout()) {
- LOG(LS_WARNING) << "Failed to resume playout";
- ret = false;
- }
- if (!channel->ResumeSend()) {
- LOG(LS_WARNING) << "Failed to resume send";
- ret = false;
- }
- }
if (ret) {
- LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name
- << ") and speaker to (id="<< out_id << " name=" << out_name
- << ")";
+ LOG(LS_INFO) << "Set microphone to (id=" << in_id
+ << ") and speaker to (id=" << out_id << ")";
}
-
- return ret;
-#else
- return true;
#endif // !IOS
}
-bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId(
- bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) {
- RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- // In Linux, VoiceEngine uses the same device dev_id as the device manager.
-#if defined(LINUX) || defined(ANDROID)
- *rtc_id = dev_id;
- return true;
-#else
- // In Windows and Mac, we need to find the VoiceEngine device id by name
- // unless the input dev_id is the default device id.
- if (kDefaultAudioDeviceId == dev_id) {
- *rtc_id = dev_id;
- return true;
- }
-
- // Get the number of VoiceEngine audio devices.
- int count = 0;
- if (is_input) {
- if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) {
- LOG_RTCERR0(GetNumOfRecordingDevices);
- return false;
- }
- } else {
- if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) {
- LOG_RTCERR0(GetNumOfPlayoutDevices);
- return false;
- }
- }
-
- for (int i = 0; i < count; ++i) {
- char name[128];
- char guid[128];
- if (is_input) {
- voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid);
- LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name;
- } else {
- voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid);
- LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name;
- }
-
- std::string webrtc_name(name);
- if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) {
- *rtc_id = i;
- return true;
- }
- }
- LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name;
- return false;
-#endif
-}
-
bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
unsigned int ulevel;
@@ -1511,19 +1369,17 @@
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
options_.SetAll(options);
-
- if (send_ != SEND_NOTHING) {
- if (!engine()->ApplyOptions(options_)) {
- LOG(LS_WARNING) <<
- "Failed to apply engine options during channel SetOptions.";
- return false;
- }
+ if (!engine()->ApplyOptions(options_)) {
+ LOG(LS_WARNING) <<
+ "Failed to apply engine options during channel SetOptions.";
+ return false;
}
if (dscp_option_changed) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
- if (options_.dscp.value_or(false))
+ if (options_.dscp.value_or(false)) {
dscp = kAudioDscpValue;
+ }
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
}
@@ -1921,7 +1777,7 @@
return true;
}
- // Apply channel specific options.
+ // Apply channel specific options when channel is enabled for sending.
if (send == SEND_MICROPHONE) {
engine()->ApplyOptions(options_);
}
@@ -1933,13 +1789,6 @@
}
}
- // Clear up the options after stopping sending. Since we may previously have
- // applied the channel specific options, now apply the original options stored
- // in WebRtcVoiceEngine.
- if (send == SEND_NOTHING) {
- engine()->ApplyOptions(engine()->GetOptions());
- }
-
send_ = send;
return true;
}
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 3222861..1de4fb9 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -29,7 +29,6 @@
#define TALK_MEDIA_WEBRTCVOICEENGINE_H_
#include <map>
-#include <set>
#include <string>
#include <vector>
@@ -72,9 +71,6 @@
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const AudioOptions& options);
- AudioOptions GetOptions() const { return options_; }
- bool SetOptions(const AudioOptions& options);
- bool SetDevices(const Device* in_device, const Device* out_device);
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
int GetInputLevel();
@@ -118,15 +114,11 @@
// ignored. This allows us to selectively turn on and off different options
// easily at any time.
bool ApplyOptions(const AudioOptions& options);
+ void SetDefaultDevices();
// webrtc::TraceCallback:
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
- // Given the device type, name, and id, find device id. Return true and
- // set the output parameter rtc_id if successful.
- bool FindWebRtcAudioDeviceId(
- bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
-
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
@@ -138,16 +130,13 @@
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_ = nullptr;
- bool is_dumping_aec_ = false;
std::vector<AudioCodec> codecs_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
- webrtc::AgcConfig default_agc_config_;
-
webrtc::Config voe_config_;
-
bool initialized_ = false;
- AudioOptions options_;
+ bool is_dumping_aec_ = false;
+ webrtc::AgcConfig default_agc_config_;
// Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
// values, and apply them in case they are missing in the audio options. We
// need to do this because SetExtraOptions() will revert to defaults for
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index f639e74..a62bcb2 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -80,7 +80,6 @@
channel_(nullptr) {
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
- options_adjust_agc_.adjust_agc_delta = rtc::Optional<int>(-10);
}
bool SetupEngine() {
if (!engine_.Init(rtc::Thread::Current())) {
@@ -398,10 +397,8 @@
cricket::FakeWebRtcVoiceEngine voe_;
cricket::WebRtcVoiceEngine engine_;
cricket::VoiceMediaChannel* channel_;
-
cricket::AudioSendParameters send_parameters_;
cricket::AudioRecvParameters recv_parameters_;
- cricket::AudioOptions options_adjust_agc_;
};
// Tests that our stub library "works".
@@ -2161,96 +2158,17 @@
EXPECT_FALSE(voe_.GetPlayout(channel_num1));
}
-// Test that we can set the devices to use.
-TEST_F(WebRtcVoiceEngineTestFake, SetDevices) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- int send_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- int recv_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
- cricket::Device default_dev(cricket::kFakeDefaultDeviceName,
- cricket::kFakeDefaultDeviceId);
- cricket::Device dev(cricket::kFakeDeviceName,
- cricket::kFakeDeviceId);
-
- // Test SetDevices() while not sending or playing.
- EXPECT_TRUE(engine_.SetDevices(&default_dev, &default_dev));
-
- // Test SetDevices() while sending and playing.
- EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
- EXPECT_TRUE(channel_->SetPlayout(true));
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-
- // Test that failure to open newly selected devices does not prevent opening
- // ones after that.
- voe_.set_playout_fail_channel(recv_channel);
- voe_.set_send_fail_channel(send_channel);
-
- EXPECT_FALSE(engine_.SetDevices(&default_dev, &default_dev));
-
- EXPECT_FALSE(voe_.GetSend(send_channel));
- EXPECT_FALSE(voe_.GetPlayout(recv_channel));
-
- voe_.set_playout_fail_channel(-1);
- voe_.set_send_fail_channel(-1);
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-}
-
-// Test that we can set the devices to use even if we failed to
-// open the initial ones.
-TEST_F(WebRtcVoiceEngineTestFake, SetDevicesWithInitiallyBadDevices) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- int send_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- int recv_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
- cricket::Device default_dev(cricket::kFakeDefaultDeviceName,
- cricket::kFakeDefaultDeviceId);
- cricket::Device dev(cricket::kFakeDeviceName,
- cricket::kFakeDeviceId);
-
- // Test that failure to open devices selected before starting
- // send/play does not prevent opening newly selected ones after that.
- voe_.set_playout_fail_channel(recv_channel);
- voe_.set_send_fail_channel(send_channel);
-
- EXPECT_TRUE(engine_.SetDevices(&default_dev, &default_dev));
-
- EXPECT_FALSE(channel_->SetSend(cricket::SEND_MICROPHONE));
- EXPECT_FALSE(channel_->SetPlayout(true));
- EXPECT_FALSE(voe_.GetSend(send_channel));
- EXPECT_FALSE(voe_.GetPlayout(recv_channel));
-
- voe_.set_playout_fail_channel(-1);
- voe_.set_send_fail_channel(-1);
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-}
-
// Test that we can create a channel configured for Codian bridges,
// and start sending on it.
TEST_F(WebRtcVoiceEngineTestFake, CodianSend) {
EXPECT_TRUE(SetupEngineWithSendStream());
+ cricket::AudioOptions options_adjust_agc;
+ options_adjust_agc.adjust_agc_delta = rtc::Optional<int>(-10);
int channel_num = voe_.GetLastChannel();
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(0, agc_config.targetLeveldBOv);
- send_parameters_.options = options_adjust_agc_;
+ send_parameters_.options = options_adjust_agc;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
EXPECT_TRUE(voe_.GetSend(channel_num));
@@ -2259,7 +2177,6 @@
EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
EXPECT_FALSE(voe_.GetSend(channel_num));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
- EXPECT_EQ(0, agc_config.targetLeveldBOv); // level was restored
}
TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
@@ -2267,14 +2184,12 @@
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(0, agc_config.targetLeveldBOv);
-
- cricket::AudioOptions options;
- options.tx_agc_target_dbov = rtc::Optional<uint16_t>(3);
- options.tx_agc_digital_compression_gain = rtc::Optional<uint16_t>(9);
- options.tx_agc_limiter = rtc::Optional<bool>(true);
- options.auto_gain_control = rtc::Optional<bool>(true);
- EXPECT_TRUE(engine_.SetOptions(options));
-
+ send_parameters_.options.tx_agc_target_dbov = rtc::Optional<uint16_t>(3);
+ send_parameters_.options.tx_agc_digital_compression_gain =
+ rtc::Optional<uint16_t>(9);
+ send_parameters_.options.tx_agc_limiter = rtc::Optional<bool>(true);
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(3, agc_config.targetLeveldBOv);
EXPECT_EQ(9, agc_config.digitalCompressionGaindB);
@@ -2282,19 +2197,18 @@
// Check interaction with adjust_agc_delta. Both should be respected, for
// backwards compatibility.
- options.adjust_agc_delta = rtc::Optional<int>(-10);
- EXPECT_TRUE(engine_.SetOptions(options));
-
+ send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(-10);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(13, agc_config.targetLeveldBOv);
}
TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
EXPECT_TRUE(SetupEngineWithSendStream());
- cricket::AudioOptions options;
- options.recording_sample_rate = rtc::Optional<uint32_t>(48000);
- options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
- EXPECT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.recording_sample_rate =
+ rtc::Optional<uint32_t>(48000);
+ send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
unsigned int recording_sample_rate, playout_sample_rate;
EXPECT_EQ(0, voe_.RecordingSampleRate(&recording_sample_rate));
@@ -2638,10 +2552,12 @@
EXPECT_TRUE(typing_detection_enabled);
EXPECT_EQ(ec_mode, webrtc::kEcConference);
EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
+ EXPECT_EQ(50, voe_.GetNetEqCapacity());
+ EXPECT_FALSE(voe_.GetNetEqFastAccelerate());
- // Nothing set, so all ignored.
- cricket::AudioOptions options;
- ASSERT_TRUE(engine_.SetOptions(options));
+ // Nothing set in AudioOptions, so everything should be as default.
+ send_parameters_.options = cricket::AudioOptions();
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2661,20 +2577,19 @@
EXPECT_TRUE(typing_detection_enabled);
EXPECT_EQ(ec_mode, webrtc::kEcConference);
EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
- EXPECT_EQ(50, voe_.GetNetEqCapacity()); // From GetDefaultEngineOptions().
- EXPECT_FALSE(
- voe_.GetNetEqFastAccelerate()); // From GetDefaultEngineOptions().
+ EXPECT_EQ(50, voe_.GetNetEqCapacity());
+ EXPECT_FALSE(voe_.GetNetEqFastAccelerate());
// Turn echo cancellation off
- options.echo_cancellation = rtc::Optional<bool>(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_FALSE(ec_enabled);
// Turn echo cancellation back on, with settings, and make sure
// nothing else changed.
- options.echo_cancellation = rtc::Optional<bool>(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2696,8 +2611,8 @@
// Turn on delay agnostic aec and make sure nothing change w.r.t. echo
// control.
- options.delay_agnostic_aec = rtc::Optional<bool>(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
EXPECT_TRUE(ec_enabled);
@@ -2705,41 +2620,41 @@
EXPECT_EQ(ec_mode, webrtc::kEcConference);
// Turn off echo cancellation and delay agnostic aec.
- options.delay_agnostic_aec = rtc::Optional<bool>(false);
- options.extended_filter_aec = rtc::Optional<bool>(false);
- options.echo_cancellation = rtc::Optional<bool>(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(false);
+ send_parameters_.options.extended_filter_aec = rtc::Optional<bool>(false);
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_FALSE(ec_enabled);
// Turning delay agnostic aec back on should also turn on echo cancellation.
- options.delay_agnostic_aec = rtc::Optional<bool>(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_TRUE(ec_enabled);
EXPECT_TRUE(voe_.ec_metrics_enabled());
EXPECT_EQ(ec_mode, webrtc::kEcConference);
// Turn off AGC
- options.auto_gain_control = rtc::Optional<bool>(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetAgcStatus(agc_enabled, agc_mode);
EXPECT_FALSE(agc_enabled);
// Turn AGC back on
- options.auto_gain_control = rtc::Optional<bool>(true);
- options.adjust_agc_delta = rtc::Optional<int>();
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true);
+ send_parameters_.options.adjust_agc_delta = rtc::Optional<int>();
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetAgcStatus(agc_enabled, agc_mode);
EXPECT_TRUE(agc_enabled);
voe_.GetAgcConfig(agc_config);
EXPECT_EQ(0, agc_config.targetLeveldBOv);
// Turn off other options (and stereo swapping on).
- options.noise_suppression = rtc::Optional<bool>(false);
- options.highpass_filter = rtc::Optional<bool>(false);
- options.typing_detection = rtc::Optional<bool>(false);
- options.stereo_swapping = rtc::Optional<bool>(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.noise_suppression = rtc::Optional<bool>(false);
+ send_parameters_.options.highpass_filter = rtc::Optional<bool>(false);
+ send_parameters_.options.typing_detection = rtc::Optional<bool>(false);
+ send_parameters_.options.stereo_swapping = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetNsStatus(ns_enabled, ns_mode);
highpass_filter_enabled = voe_.IsHighPassFilterEnabled();
stereo_swapping_enabled = voe_.IsStereoChannelSwappingEnabled();
@@ -2750,7 +2665,7 @@
EXPECT_TRUE(stereo_swapping_enabled);
// Set options again to ensure it has no impact.
- ASSERT_TRUE(engine_.SetOptions(options));
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetNsStatus(ns_enabled, ns_mode);
EXPECT_TRUE(ec_enabled);
@@ -2850,7 +2765,7 @@
expected_options.noise_suppression = rtc::Optional<bool>(true);
EXPECT_EQ(expected_options, channel2->options());
- ASSERT_TRUE(engine_.SetOptions(parameters_options_all.options));
+ ASSERT_TRUE(channel_->SetSendParameters(parameters_options_all));
bool ec_enabled;
webrtc::EcModes ec_mode;
bool agc_enabled;
@@ -2872,14 +2787,6 @@
EXPECT_TRUE(agc_enabled);
EXPECT_FALSE(ns_enabled);
- channel1->SetSend(cricket::SEND_NOTHING);
- voe_.GetEcStatus(ec_enabled, ec_mode);
- voe_.GetAgcStatus(agc_enabled, agc_mode);
- voe_.GetNsStatus(ns_enabled, ns_mode);
- EXPECT_TRUE(ec_enabled);
- EXPECT_TRUE(agc_enabled);
- EXPECT_TRUE(ns_enabled);
-
channel2->SetSend(cricket::SEND_MICROPHONE);
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2888,16 +2795,8 @@
EXPECT_FALSE(agc_enabled);
EXPECT_TRUE(ns_enabled);
- channel2->SetSend(cricket::SEND_NOTHING);
- voe_.GetEcStatus(ec_enabled, ec_mode);
- voe_.GetAgcStatus(agc_enabled, agc_mode);
- voe_.GetNsStatus(ns_enabled, ns_mode);
- EXPECT_TRUE(ec_enabled);
- EXPECT_TRUE(agc_enabled);
- EXPECT_TRUE(ns_enabled);
-
// Make sure settings take effect while we are sending.
- ASSERT_TRUE(engine_.SetOptions(parameters_options_all.options));
+ ASSERT_TRUE(channel_->SetSendParameters(parameters_options_all));
cricket::AudioSendParameters parameters_options_no_agc_nor_ns =
send_parameters_;
parameters_options_no_agc_nor_ns.options.auto_gain_control =
@@ -3072,13 +2971,12 @@
send_parameters_.options.combined_audio_video_bwe = rtc::Optional<bool>(true);
EXPECT_TRUE(media_channel->SetSendParameters(send_parameters_));
- static const uint32_t kSsrcs[] = {1, 2, 3, 4};
- for (unsigned int i = 0; i < arraysize(kSsrcs); ++i) {
+ for (uint32_t ssrc : kSsrcs4) {
EXPECT_TRUE(media_channel->AddRecvStream(
- cricket::StreamParams::CreateLegacy(kSsrcs[i])));
- EXPECT_NE(nullptr, call_.GetAudioReceiveStream(kSsrcs[i]));
+ cricket::StreamParams::CreateLegacy(ssrc)));
+ EXPECT_NE(nullptr, call_.GetAudioReceiveStream(ssrc));
}
- EXPECT_EQ(arraysize(kSsrcs), call_.GetAudioReceiveStreams().size());
+ EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioReceiveStreams().size());
}
// TODO(solenberg): Remove, once recv streams are configured through Call.
@@ -3207,18 +3105,6 @@
EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), -1);
}
-// Tests for the actual WebRtc VoE library.
-
-TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
- cricket::WebRtcVoiceEngine engine;
- cricket::AudioOptions options = engine.GetOptions();
- // The default options should have at least a few things set. We purposefully
- // don't check the option values here, though.
- EXPECT_TRUE(options.echo_cancellation);
- EXPECT_TRUE(options.auto_gain_control);
- EXPECT_TRUE(options.noise_suppression);
-}
-
// Tests that the library initializes and shuts down properly.
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
cricket::WebRtcVoiceEngine engine;
diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc
index bb6fb25..eb869dd 100644
--- a/talk/session/media/channelmanager.cc
+++ b/talk/session/media/channelmanager.cc
@@ -101,8 +101,6 @@
initialized_ = false;
main_thread_ = rtc::Thread::Current();
worker_thread_ = worker_thread;
- // Get the default audio options from the media engine.
- audio_options_ = media_engine_->GetAudioOptions();
audio_output_volume_ = kNotSetOutputVolume;
local_renderer_ = NULL;
capturing_ = false;
@@ -205,11 +203,6 @@
return false;
}
- if (!SetAudioOptions(audio_options_)) {
- LOG(LS_WARNING) << "Failed to SetAudioOptions with options: "
- << audio_options_.ToString();
- }
-
// If audio_output_volume_ has been set via SetOutputVolume(), set the
// audio output volume of the engine.
if (kNotSetOutputVolume != audio_output_volume_ &&
@@ -427,43 +420,6 @@
delete data_channel;
}
-bool ChannelManager::SetAudioOptions(const AudioOptions& options) {
- // "Get device ids from DeviceManager" - these are the defaults returned.
- Device in_dev("", -1);
- Device out_dev("", -1);
-
- // If we're initialized, pass the settings to the media engine.
- bool ret = true;
- if (initialized_) {
- ret = worker_thread_->Invoke<bool>(
- Bind(&ChannelManager::SetAudioOptions_w, this,
- options, &in_dev, &out_dev));
- }
-
- // If all worked well, save the values for use in GetAudioOptions.
- if (ret) {
- audio_options_ = options;
- }
- return ret;
-}
-
-bool ChannelManager::SetAudioOptions_w(
- const AudioOptions& options,
- const Device* in_dev, const Device* out_dev) {
- ASSERT(worker_thread_ == rtc::Thread::Current());
- ASSERT(initialized_);
-
- // Set audio options
- bool ret = media_engine_->SetAudioOptions(options);
-
- // Set the audio devices
- if (ret) {
- ret = media_engine_->SetSoundDevices(in_dev, out_dev);
- }
-
- return ret;
-}
-
bool ChannelManager::GetOutputVolume(int* level) {
if (!initialized_) {
return false;
diff --git a/talk/session/media/channelmanager.h b/talk/session/media/channelmanager.h
index d333dea..a141fe1 100644
--- a/talk/session/media/channelmanager.h
+++ b/talk/session/media/channelmanager.h
@@ -177,11 +177,6 @@
sigslot::signal2<VideoCapturer*, CaptureState> SignalVideoCaptureStateChange;
- protected:
- // Adds non-transient parameters which can only be changed through the
- // options store.
- bool SetAudioOptions(const AudioOptions& options);
-
private:
typedef std::vector<VoiceChannel*> VoiceChannels;
typedef std::vector<VideoChannel*> VideoChannels;
@@ -213,8 +208,6 @@
bool rtcp,
DataChannelType data_channel_type);
void DestroyDataChannel_w(DataChannel* data_channel);
- bool SetAudioOptions_w(const AudioOptions& options,
- const Device* in_dev, const Device* out_dev);
void OnVideoCaptureStateChange(VideoCapturer* capturer,
CaptureState result);
void GetSupportedFormats_w(
@@ -234,7 +227,6 @@
VideoChannels video_channels_;
DataChannels data_channels_;
- AudioOptions audio_options_;
int audio_output_volume_;
VideoEncoderConfig default_video_encoder_config_;
VideoRenderer* local_renderer_;
