Update libjingle to 57692857
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 4911c59..1b77671 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -230,6 +230,7 @@
options.experimental_agc.Set(false);
options.experimental_aec.Set(false);
options.aec_dump.Set(false);
+ options.experimental_acm.Set(false);
return options;
}
@@ -260,7 +261,7 @@
if (!engine_->voe_sc()) {
return false;
}
- webrtc_channel_ = engine_->voe_sc()->base()->CreateChannel();
+ webrtc_channel_ = engine_->CreateSoundclipVoiceChannel();
if (webrtc_channel_ == -1) {
LOG_RTCERR0(CreateChannel);
return false;
@@ -333,6 +334,7 @@
log_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
+ use_experimental_acm_(false),
tx_processor_ssrc_(0),
rx_processor_ssrc_(0) {
Construct();
@@ -350,6 +352,7 @@
log_filter_(SeverityToFilter(kDefaultLogSeverity)),
is_dumping_aec_(false),
desired_local_monitor_enable_(false),
+ use_experimental_acm_(false),
tx_processor_ssrc_(0),
rx_processor_ssrc_(0) {
Construct();
@@ -377,6 +380,10 @@
RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
kRtpAudioLevelHeaderExtensionId));
options_ = GetDefaultEngineOptions();
+
+ // Initialize the VoE Configuration to the default ACM.
+ voe_config_.Set<webrtc::AudioCodingModuleFactory>(
+ new webrtc::AudioCodingModuleFactory);
}
static bool IsOpus(const AudioCodec& codec) {
@@ -714,6 +721,12 @@
LOG(LS_INFO) << "Applying audio options: " << options.ToString();
+ // Configure whether ACM1 or ACM2 is used.
+ bool enable_acm2 = false;
+ if (options.experimental_acm.Get(&enable_acm2)) {
+ EnableExperimentalAcm(enable_acm2);
+ }
+
webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
bool echo_cancellation;
@@ -940,7 +953,7 @@
}
if (ret) {
if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
- LOG_RTCERR2(SetRecordingDevice, in_device->name, in_id);
+ LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
ret = false;
}
}
@@ -952,7 +965,7 @@
}
if (ret) {
if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
- LOG_RTCERR2(SetPlayoutDevice, out_device->name, out_id);
+ LOG_RTCERR2(SetPlayoutDevice, out_name, out_id);
ret = false;
}
}
@@ -1248,6 +1261,21 @@
return false;
}
+void WebRtcVoiceEngine::EnableExperimentalAcm(bool enable) {
+ if (enable == use_experimental_acm_)
+ return;
+ if (enable) {
+ LOG(LS_INFO) << "VoiceEngine is set to use new ACM (ACM2 + NetEq4).";
+ voe_config_.Set<webrtc::AudioCodingModuleFactory>(
+ new webrtc::NewAudioCodingModuleFactory());
+ } else {
+ LOG(LS_INFO) << "VoiceEngine is set to use legacy ACM (ACM1 + Neteq3).";
+ voe_config_.Set<webrtc::AudioCodingModuleFactory>(
+ new webrtc::AudioCodingModuleFactory());
+ }
+ use_experimental_acm_ = enable;
+}
+
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
talk_base::LoggingSeverity sev = talk_base::LS_VERBOSE;
@@ -1580,6 +1608,22 @@
}
}
+int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) {
+#ifdef USE_WEBRTC_DEV_BRANCH
+ return voice_engine_wrapper->base()->CreateChannel(voe_config_);
+#else
+ return voice_engine_wrapper->base()->CreateChannel();
+#endif
+}
+
+int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
+ return CreateVoiceChannel(voe_wrapper_.get());
+}
+
+int WebRtcVoiceEngine::CreateSoundclipVoiceChannel() {
+ return CreateVoiceChannel(voe_wrapper_sc_.get());
+}
+
// This struct relies on the generated copy constructor and assignment operator
// since it is used in an stl::map.
struct WebRtcVoiceMediaChannel::WebRtcVoiceChannelInfo {
@@ -1597,7 +1641,7 @@
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
- engine->voe()->base()->CreateChannel()),
+ engine->CreateMediaVoiceChannel()),
send_bw_setting_(false),
send_autobw_(false),
send_bw_bps_(0),
@@ -2231,7 +2275,7 @@
channel = voe_channel();
} else {
// Create a new channel for sending audio data.
- channel = engine()->voe()->base()->CreateChannel();
+ channel = engine()->CreateMediaVoiceChannel();
if (channel == -1) {
LOG_RTCERR0(CreateChannel);
return false;
@@ -2346,7 +2390,7 @@
}
// Create a new channel for receiving audio data.
- int channel = engine()->voe()->base()->CreateChannel();
+ int channel = engine()->CreateMediaVoiceChannel();
if (channel == -1) {
LOG_RTCERR0(CreateChannel);
return false;
@@ -2966,7 +3010,7 @@
continue;
}
- sinfo.ssrc = ssrc;
+ sinfo.add_ssrc(ssrc);
sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
sinfo.bytes_sent = cs.bytesSent;
sinfo.packets_sent = cs.packetsSent;
@@ -2988,7 +3032,7 @@
for (iter = receive_blocks.begin(); iter != receive_blocks.end();
++iter) {
// Lookup report for send ssrc only.
- if (iter->source_SSRC == sinfo.ssrc) {
+ if (iter->source_SSRC == sinfo.ssrc()) {
// Convert Q8 to floating point.
sinfo.fraction_lost = static_cast<float>(iter->fraction_lost) / 256;
// Convert samples to milliseconds.
@@ -3041,7 +3085,7 @@
engine()->voe()->rtp()->GetRTCPStatistics(*it, cs) != -1 &&
engine()->voe()->codec()->GetRecCodec(*it, codec) != -1) {
VoiceReceiverInfo rinfo;
- rinfo.ssrc = ssrc;
+ rinfo.add_ssrc(ssrc);
rinfo.bytes_rcvd = cs.bytesReceived;
rinfo.packets_rcvd = cs.packetsReceived;
// The next four fields are from the most recently sent RTCP report.
