Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49329004
Cr-Commit-Position: refs/heads/master@{#9169}
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index 460ad7e..f8f363a 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -1304,6 +1304,9 @@
"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 =
@@ -1312,6 +1315,8 @@
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 8fcc975..80e7bfe 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
@@ -128,12 +128,14 @@
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 39284c1..5226041 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -382,9 +382,7 @@
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), constraints,
- dtls_identity_service,
- configuration.type,
- configuration.bundle_policy))
+ dtls_identity_service, configuration))
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 ed4f5b3..e32676e 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -211,11 +211,13 @@
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) {}
+ tcp_candidate_policy(kTcpCandidatePolicyEnabled),
+ audio_jitter_buffer_max_packets(50) {}
};
struct RTCOfferAnswerOptions {
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index dd5d858..e2a9d60 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -521,9 +521,8 @@
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints,
DTLSIdentityServiceInterface* dtls_identity_service,
- PeerConnectionInterface::IceTransportsType ice_transport_type,
- PeerConnectionInterface::BundlePolicy bundle_policy) {
- bundle_policy_ = bundle_policy;
+ const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
+ bundle_policy_ = rtc_configuration.bundle_policy;
// TODO(perkj): Take |constraints| into consideration. Return false if not all
// mandatory constraints can be fulfilled. Note that |constraints|
@@ -640,6 +639,9 @@
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,
@@ -667,7 +669,7 @@
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
port_allocator()->set_candidate_filter(
- ConvertIceTransportTypeToCandidateFilter(ice_transport_type));
+ ConvertIceTransportTypeToCandidateFilter(rtc_configuration.type));
return true;
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index ce0fb0c..aa1deb5 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,
- PeerConnectionInterface::IceTransportsType ice_transport_type,
- PeerConnectionInterface::BundlePolicy bundle_policy);
+ bool Initialize(
+ const PeerConnectionFactoryInterface::Options& options,
+ const MediaConstraintsInterface* constraints,
+ DTLSIdentityServiceInterface* dtls_identity_service,
+ const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
// 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 3efc112..e4f39f8 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -156,6 +156,8 @@
"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,
@@ -383,8 +385,7 @@
void Init(
DTLSIdentityServiceInterface* identity_service,
- PeerConnectionInterface::IceTransportsType ice_transport_type,
- PeerConnectionInterface::BundlePolicy bundle_policy) {
+ const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
ASSERT_TRUE(session_.get() == NULL);
session_.reset(new WebRtcSessionForTest(
channel_manager_.get(), rtc::Thread::Current(),
@@ -398,33 +399,51 @@
observer_.ice_gathering_state_);
EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
- identity_service, ice_transport_type,
- bundle_policy));
+ identity_service, rtc_configuration));
session_->set_metrics_observer(&metrics_observer_);
}
void Init() {
- Init(NULL, PeerConnectionInterface::kAll,
- PeerConnectionInterface::kBundlePolicyBalanced);
+ PeerConnectionInterface::RTCConfiguration configuration;
+ configuration.type = PeerConnectionInterface::kAll;
+ configuration.bundle_policy =
+ PeerConnectionInterface::kBundlePolicyBalanced;
+ configuration.audio_jitter_buffer_max_packets =
+ kAudioJitterBufferMaxPackets;
+ Init(NULL, configuration);
}
void InitWithIceTransport(
PeerConnectionInterface::IceTransportsType ice_transport_type) {
- Init(NULL, ice_transport_type,
- PeerConnectionInterface::kBundlePolicyBalanced);
+ PeerConnectionInterface::RTCConfiguration configuration;
+ configuration.type = ice_transport_type;
+ configuration.bundle_policy =
+ PeerConnectionInterface::kBundlePolicyBalanced;
+ configuration.audio_jitter_buffer_max_packets =
+ kAudioJitterBufferMaxPackets;
+ Init(NULL, configuration);
}
void InitWithBundlePolicy(
PeerConnectionInterface::BundlePolicy bundle_policy) {
- Init(NULL, PeerConnectionInterface::kAll, bundle_policy);
+ PeerConnectionInterface::RTCConfiguration configuration;
+ configuration.type = PeerConnectionInterface::kAll;
+ configuration.bundle_policy = bundle_policy;
+ configuration.audio_jitter_buffer_max_packets =
+ kAudioJitterBufferMaxPackets;
+ Init(NULL, configuration);
}
void InitWithDtls(bool identity_request_should_fail = false) {
FakeIdentityService* identity_service = new FakeIdentityService();
identity_service->set_should_fail(identity_request_should_fail);
- Init(identity_service,
- PeerConnectionInterface::kAll,
- PeerConnectionInterface::kBundlePolicyBalanced);
+ PeerConnectionInterface::RTCConfiguration configuration;
+ configuration.type = PeerConnectionInterface::kAll;
+ configuration.bundle_policy =
+ PeerConnectionInterface::kBundlePolicyBalanced;
+ configuration.audio_jitter_buffer_max_packets =
+ kAudioJitterBufferMaxPackets;
+ Init(identity_service, configuration);
}
void InitWithDtmfCodec() {
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index dd78f0e..d77ddbb 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -150,6 +150,8 @@
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);
@@ -180,6 +182,7 @@
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 &&
@@ -210,6 +213,8 @@
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);
@@ -248,6 +253,8 @@
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 24ef846..5b86b53 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -41,6 +41,7 @@
#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 {
@@ -213,7 +214,8 @@
send_audio_level_ext_(-1),
receive_audio_level_ext_(-1),
send_absolute_sender_time_ext_(-1),
- receive_absolute_sender_time_ext_(-1) {
+ receive_absolute_sender_time_ext_(-1),
+ neteq_capacity(-1) {
memset(&send_codec, 0, sizeof(send_codec));
memset(&rx_agc_config, 0, sizeof(rx_agc_config));
}
@@ -249,6 +251,7 @@
webrtc::CodecInst send_codec;
webrtc::PacketTime last_rtp_packet_time;
std::list<std::string> packets;
+ int neteq_capacity;
};
FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs,
@@ -391,7 +394,7 @@
true);
}
}
- int AddChannel() {
+ int AddChannel(const webrtc::Config& config) {
if (fail_create_channel_) {
return -1;
}
@@ -401,6 +404,9 @@
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_;
}
@@ -447,10 +453,11 @@
return &audio_processing_;
}
WEBRTC_FUNC(CreateChannel, ()) {
- return AddChannel();
+ webrtc::Config empty_config;
+ return AddChannel(empty_config);
}
- WEBRTC_FUNC(CreateChannel, (const webrtc::Config& /*config*/)) {
- return AddChannel();
+ WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
+ return AddChannel(config);
}
WEBRTC_FUNC(DeleteChannel, (int channel)) {
WEBRTC_CHECK_CHANNEL(channel);
@@ -1243,6 +1250,11 @@
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 baae3de..fcf8ef7 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -353,6 +353,7 @@
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);
@@ -955,6 +956,14 @@
}
}
+ 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 4596c4d..242467d 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -46,6 +46,7 @@
#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 7737f34..f995093 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -2882,6 +2882,7 @@
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 4e2faa3..09633ed 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -113,6 +113,20 @@
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 fa63491..51b9a78 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -10,6 +10,7 @@
#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"
@@ -20,13 +21,20 @@
// Create module
AudioCodingModule* AudioCodingModule::Create(int id) {
- return Create(id, Clock::GetRealTimeClock());
+ Config config;
+ config.id = id;
+ config.clock = Clock::GetRealTimeClock();
+ return Create(config);
}
AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
- AudioCodingModule::Config config;
+ 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 888eb87..9cd2cb4 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -99,6 +99,7 @@
//
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 d00ddf3..c3fba0d 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -10,9 +10,12 @@
#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"
@@ -757,8 +760,6 @@
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(),
@@ -828,6 +829,16 @@
{
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();
