Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55369004
Cr-Commit-Position: refs/heads/master@{#9165}
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index f8f363a..460ad7e 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -1304,9 +1304,6 @@
"Ljava/util/List;");
jobject j_ice_servers = GetObjectField(jni, j_rtc_config, j_ice_servers_id);
- jfieldID j_audio_jitter_buffer_max_packets_id = GetFieldID(
- jni, j_rtc_config_class, "audioJitterBufferMaxPackets",
- "I");
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.type =
@@ -1315,8 +1312,6 @@
rtc_config.tcp_candidate_policy =
JavaTcpCandidatePolicyToNativeType(jni, j_tcp_candidate_policy);
JavaIceServersToJsepIceServers(jni, j_ice_servers, &rtc_config.servers);
- rtc_config.audio_jitter_buffer_max_packets =
- GetIntField(jni, j_rtc_config, j_audio_jitter_buffer_max_packets_id);
PCOJava* observer = reinterpret_cast<PCOJava*>(observer_p);
observer->SetConstraints(new ConstraintsWrapper(jni, j_constraints));
diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
index 80e7bfe..8fcc975 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
@@ -128,14 +128,12 @@
public List<IceServer> iceServers;
public BundlePolicy bundlePolicy;
public TcpCandidatePolicy tcpCandidatePolicy;
- public int audioJitterBufferMaxPackets;
public RTCConfiguration(List<IceServer> iceServers) {
iceTransportsType = IceTransportsType.ALL;
bundlePolicy = BundlePolicy.BALANCED;
tcpCandidatePolicy = TcpCandidatePolicy.ENABLED;
this.iceServers = iceServers;
- audioJitterBufferMaxPackets = 50;
}
};
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 5226041..39284c1 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -382,7 +382,9 @@
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), constraints,
- dtls_identity_service, configuration))
+ dtls_identity_service,
+ configuration.type,
+ configuration.bundle_policy))
return false;
// Register PeerConnection as receiver of local ice candidates.
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index e32676e..ed4f5b3 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -211,13 +211,11 @@
IceServers servers;
BundlePolicy bundle_policy;
TcpCandidatePolicy tcp_candidate_policy;
- int audio_jitter_buffer_max_packets;
RTCConfiguration()
: type(kAll),
bundle_policy(kBundlePolicyBalanced),
- tcp_candidate_policy(kTcpCandidatePolicyEnabled),
- audio_jitter_buffer_max_packets(50) {}
+ tcp_candidate_policy(kTcpCandidatePolicyEnabled) {}
};
struct RTCOfferAnswerOptions {
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index e2a9d60..dd5d858 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -521,8 +521,9 @@
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints,
DTLSIdentityServiceInterface* dtls_identity_service,
- const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
- bundle_policy_ = rtc_configuration.bundle_policy;
+ PeerConnectionInterface::IceTransportsType ice_transport_type,
+ PeerConnectionInterface::BundlePolicy bundle_policy) {
+ bundle_policy_ = bundle_policy;
// TODO(perkj): Take |constraints| into consideration. Return false if not all
// mandatory constraints can be fulfilled. Note that |constraints|
@@ -639,9 +640,6 @@
MediaConstraintsInterface::kCombinedAudioVideoBwe,
&audio_options_.combined_audio_video_bwe);
- audio_options_.audio_jitter_buffer_max_packets.Set(
- rtc_configuration.audio_jitter_buffer_max_packets);
-
const cricket::VideoCodec default_codec(
JsepSessionDescription::kDefaultVideoCodecId,
JsepSessionDescription::kDefaultVideoCodecName,
@@ -669,7 +667,7 @@
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
port_allocator()->set_candidate_filter(
- ConvertIceTransportTypeToCandidateFilter(rtc_configuration.type));
+ ConvertIceTransportTypeToCandidateFilter(ice_transport_type));
return true;
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index aa1deb5..ce0fb0c 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -117,11 +117,11 @@
MediaStreamSignaling* mediastream_signaling);
virtual ~WebRtcSession();
- bool Initialize(
- const PeerConnectionFactoryInterface::Options& options,
- const MediaConstraintsInterface* constraints,
- DTLSIdentityServiceInterface* dtls_identity_service,
- const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
+ bool Initialize(const PeerConnectionFactoryInterface::Options& options,
+ const MediaConstraintsInterface* constraints,
+ DTLSIdentityServiceInterface* dtls_identity_service,
+ PeerConnectionInterface::IceTransportsType ice_transport_type,
+ PeerConnectionInterface::BundlePolicy bundle_policy);
// Deletes the voice, video and data channel and changes the session state
// to STATE_RECEIVEDTERMINATE.
void Terminate();
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index e4f39f8..3efc112 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -156,8 +156,6 @@
"a=rtpmap:96 rtx/90000\r\n"
"a=fmtp:96 apt=0\r\n";
-static const int kAudioJitterBufferMaxPackets = 50;
-
// Add some extra |newlines| to the |message| after |line|.
static void InjectAfter(const std::string& line,
const std::string& newlines,
@@ -385,7 +383,8 @@
void Init(
DTLSIdentityServiceInterface* identity_service,
- const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
+ PeerConnectionInterface::IceTransportsType ice_transport_type,
+ PeerConnectionInterface::BundlePolicy bundle_policy) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
channel_manager_.get(), rtc::Thread::Current(),
@@ -399,51 +398,33 @@
observer_.ice_gathering_state_);
EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
- identity_service, rtc_configuration));
+ identity_service, ice_transport_type,
+ bundle_policy));
session_->set_metrics_observer(&metrics_observer_);
}
void Init() {
- PeerConnectionInterface::RTCConfiguration configuration;
- configuration.type = PeerConnectionInterface::kAll;
- configuration.bundle_policy =
- PeerConnectionInterface::kBundlePolicyBalanced;
- configuration.audio_jitter_buffer_max_packets =
- kAudioJitterBufferMaxPackets;
- Init(NULL, configuration);
+ Init(NULL, PeerConnectionInterface::kAll,
+ PeerConnectionInterface::kBundlePolicyBalanced);
}
void InitWithIceTransport(
PeerConnectionInterface::IceTransportsType ice_transport_type) {
- PeerConnectionInterface::RTCConfiguration configuration;
- configuration.type = ice_transport_type;
- configuration.bundle_policy =
- PeerConnectionInterface::kBundlePolicyBalanced;
- configuration.audio_jitter_buffer_max_packets =
- kAudioJitterBufferMaxPackets;
- Init(NULL, configuration);
+ Init(NULL, ice_transport_type,
+ PeerConnectionInterface::kBundlePolicyBalanced);
}
void InitWithBundlePolicy(
PeerConnectionInterface::BundlePolicy bundle_policy) {
- PeerConnectionInterface::RTCConfiguration configuration;
- configuration.type = PeerConnectionInterface::kAll;
- configuration.bundle_policy = bundle_policy;
- configuration.audio_jitter_buffer_max_packets =
- kAudioJitterBufferMaxPackets;
- Init(NULL, configuration);
+ Init(NULL, PeerConnectionInterface::kAll, bundle_policy);
}
void InitWithDtls(bool identity_request_should_fail = false) {
FakeIdentityService* identity_service = new FakeIdentityService();
identity_service->set_should_fail(identity_request_should_fail);
- PeerConnectionInterface::RTCConfiguration configuration;
- configuration.type = PeerConnectionInterface::kAll;
- configuration.bundle_policy =
- PeerConnectionInterface::kBundlePolicyBalanced;
- configuration.audio_jitter_buffer_max_packets =
- kAudioJitterBufferMaxPackets;
- Init(identity_service, configuration);
+ Init(identity_service,
+ PeerConnectionInterface::kAll,
+ PeerConnectionInterface::kBundlePolicyBalanced);
}
void InitWithDtmfCodec() {
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index d77ddbb..dd78f0e 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -150,8 +150,6 @@
noise_suppression.SetFrom(change.noise_suppression);
highpass_filter.SetFrom(change.highpass_filter);
stereo_swapping.SetFrom(change.stereo_swapping);
- audio_jitter_buffer_max_packets.SetFrom(
- change.audio_jitter_buffer_max_packets);
typing_detection.SetFrom(change.typing_detection);
aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
conference_mode.SetFrom(change.conference_mode);
@@ -182,7 +180,6 @@
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
- audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
typing_detection == o.typing_detection &&
aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
conference_mode == o.conference_mode &&
@@ -213,8 +210,6 @@
ost << ToStringIfSet("ns", noise_suppression);
ost << ToStringIfSet("hf", highpass_filter);
ost << ToStringIfSet("swap", stereo_swapping);
- ost << ToStringIfSet("audio_jitter_buffer_max_packets",
- audio_jitter_buffer_max_packets);
ost << ToStringIfSet("typing", typing_detection);
ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
ost << ToStringIfSet("conference", conference_mode);
@@ -253,8 +248,6 @@
Settable<bool> highpass_filter;
// Audio processing to swap the left and right channels.
Settable<bool> stereo_swapping;
- // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
- Settable<int> audio_jitter_buffer_max_packets;
// Audio processing to detect typing.
Settable<bool> typing_detection;
Settable<bool> aecm_generate_comfort_noise;
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 5b86b53..24ef846 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -41,7 +41,6 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
-#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace cricket {
@@ -214,8 +213,7 @@
send_audio_level_ext_(-1),
receive_audio_level_ext_(-1),
send_absolute_sender_time_ext_(-1),
- receive_absolute_sender_time_ext_(-1),
- neteq_capacity(-1) {
+ receive_absolute_sender_time_ext_(-1) {
memset(&send_codec, 0, sizeof(send_codec));
memset(&rx_agc_config, 0, sizeof(rx_agc_config));
}
@@ -251,7 +249,6 @@
webrtc::CodecInst send_codec;
webrtc::PacketTime last_rtp_packet_time;
std::list<std::string> packets;
- int neteq_capacity;
};
FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs,
@@ -394,7 +391,7 @@
true);
}
}
- int AddChannel(const webrtc::Config& config) {
+ int AddChannel() {
if (fail_create_channel_) {
return -1;
}
@@ -404,9 +401,6 @@
GetCodec(i, codec);
ch->recv_codecs.push_back(codec);
}
- if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
- ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
- }
channels_[++last_channel_] = ch;
return last_channel_;
}
@@ -453,11 +447,10 @@
return &audio_processing_;
}
WEBRTC_FUNC(CreateChannel, ()) {
- webrtc::Config empty_config;
- return AddChannel(empty_config);
+ return AddChannel();
}
- WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
- return AddChannel(config);
+ WEBRTC_FUNC(CreateChannel, (const webrtc::Config& /*config*/)) {
+ return AddChannel();
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
@@ -1250,11 +1243,6 @@
WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
webrtc::AudioFrame* frame));
WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
- int GetNetEqCapacity() const {
- auto ch = channels_.find(last_channel_);
- ASSERT(ch != channels_.end());
- return ch->second->neteq_capacity;
- }
private:
int GetNumDevices(int& num) {
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index fcf8ef7..baae3de 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -353,7 +353,6 @@
options.noise_suppression.Set(true);
options.highpass_filter.Set(true);
options.stereo_swapping.Set(false);
- options.audio_jitter_buffer_max_packets.Set(50);
options.typing_detection.Set(true);
options.conference_mode.Set(false);
options.adjust_agc_delta.Set(0);
@@ -956,14 +955,6 @@
}
}
- int audio_jitter_buffer_max_packets;
- if (options.audio_jitter_buffer_max_packets.Get(
- &audio_jitter_buffer_max_packets)) {
- LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets;
- voe_config_.Set<webrtc::NetEqCapacityConfig>(
- new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
- }
-
bool typing_detection;
if (options.typing_detection.Get(&typing_detection)) {
LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 242467d..4596c4d 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -46,7 +46,6 @@
#include "webrtc/base/thread_checker.h"
#include "webrtc/call.h"
#include "webrtc/common.h"
-#include "webrtc/config.h"
#if !defined(LIBPEERCONNECTION_LIB) && \
!defined(LIBPEERCONNECTION_IMPLEMENTATION)
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index f995093..7737f34 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -2882,7 +2882,6 @@
EXPECT_TRUE(typing_detection_enabled);
EXPECT_EQ(ec_mode, webrtc::kEcConference);
EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
- EXPECT_EQ(50, voe_.GetNetEqCapacity()); // From GetDefaultEngineOptions().
// Turn echo cancellation off
options.echo_cancellation.Set(false);
diff --git a/webrtc/config.h b/webrtc/config.h
index 09633ed..4e2faa3 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -113,20 +113,6 @@
int min_transmit_bitrate_bps;
};
-// Controls the capacity of the packet buffer in NetEq. The capacity is the
-// maximum number of packets that the buffer can contain. If the limit is
-// exceeded, the buffer will be flushed. The capacity does not affect the actual
-// audio delay in the general case, since this is governed by the target buffer
-// level (calculated from the jitter profile). It is only in the rare case of
-// severe network freezes that a higher capacity will lead to a (transient)
-// increase in audio delay.
-struct NetEqCapacityConfig {
- NetEqCapacityConfig() : enabled(false), capacity(0) {}
- explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {}
- bool enabled;
- int capacity;
-};
-
} // namespace webrtc
#endif // WEBRTC_CONFIG_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index 51b9a78..fa63491 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -10,7 +10,6 @@
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
@@ -21,20 +20,13 @@
// Create module
AudioCodingModule* AudioCodingModule::Create(int id) {
- Config config;
- config.id = id;
- config.clock = Clock::GetRealTimeClock();
- return Create(config);
+ return Create(id, Clock::GetRealTimeClock());
}
AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
- Config config;
+ AudioCodingModule::Config config;
config.id = id;
config.clock = clock;
- return Create(config);
-}
-
-AudioCodingModule* AudioCodingModule::Create(const Config& config) {
return new acm2::AudioCodingModuleImpl(config);
}
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 4660589..4d4a6f3 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -99,7 +99,6 @@
//
static AudioCodingModule* Create(int id);
static AudioCodingModule* Create(int id, Clock* clock);
- static AudioCodingModule* Create(const Config& config);
virtual ~AudioCodingModule() {};
///////////////////////////////////////////////////////////////////////////
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index ba098d9..40a974b 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -10,12 +10,9 @@
#include "webrtc/voice_engine/channel.h"
-#include <algorithm>
-
#include "webrtc/base/format_macros.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common.h"
-#include "webrtc/config.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
@@ -760,6 +757,8 @@
VoEModuleId(instanceId, channelId), Clock::GetRealTimeClock(), this,
this, this, rtp_payload_registry_.get())),
telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
+ audio_coding_(AudioCodingModule::Create(
+ VoEModuleId(instanceId, channelId))),
_rtpDumpIn(*RtpDump::CreateRtpDump()),
_rtpDumpOut(*RtpDump::CreateRtpDump()),
_outputAudioLevel(),
@@ -829,16 +828,6 @@
{
WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
"Channel::Channel() - ctor");
- AudioCodingModule::Config acm_config;
- acm_config.id = VoEModuleId(instanceId, channelId);
- if (config.Get<NetEqCapacityConfig>().enabled) {
- // Clamping the buffer capacity at 20 packets. While going lower will
- // probably work, it makes little sense.
- acm_config.neteq_config.max_packets_in_buffer =
- std::max(20, config.Get<NetEqCapacityConfig>().capacity);
- }
- audio_coding_.reset(AudioCodingModule::Create(acm_config));
-
_inbandDtmfQueue.ResetDtmf();
_inbandDtmfGenerator.Init();
_outputAudioLevel.Clear();
