Renaming bandwidth to bitrate in webrtcvoiceengine.
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.
This is to remove the confusion inside webrtcvoiceengine
BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5f2de4a..de53893 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -495,7 +495,7 @@
ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
LOG(LS_INFO) << ToString(codec);
if (IsIsac(codec)) {
- // Indicate auto-bandwidth in signaling.
+ // Indicate auto-bitrate in signaling.
codec.bitrate = 0;
}
if (IsOpus(codec)) {
@@ -1227,7 +1227,7 @@
// Apply codec-specific settings.
if (IsIsac(codec)) {
// If ISAC and an explicit bitrate is not specified,
- // enable auto bandwidth adjustment.
+ // enable auto bitrate adjustment.
voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
}
*out = voe_codec;
@@ -1792,8 +1792,8 @@
: WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine>(
engine,
engine->CreateMediaVoiceChannel()),
- send_bw_setting_(false),
- send_bw_bps_(0),
+ send_bitrate_setting_(false),
+ send_bitrate_bps_(0),
options_(),
dtmf_allowed_(false),
desired_playout_(false),
@@ -2028,9 +2028,7 @@
bool nack_enabled = nack_enabled_;
bool enable_codec_fec = false;
- // max_playback_rate <= 0 will not trigger setting of maximum encoding
- // bandwidth.
- int max_playback_rate = 0;
+ int opus_max_playback_rate = 0;
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
@@ -2048,7 +2046,6 @@
continue;
}
-
// We'll use the first codec in the list to actually send audio data.
// Be sure to use the payload type requested by the remote side.
// "red", for RED audio, is a special case where the actual codec to be
@@ -2080,7 +2077,8 @@
// For Opus as the send codec, we are to enable inband FEC if requested
// and set maximum playback rate.
if (IsOpus(*it)) {
- GetOpusConfig(*it, &send_codec, &enable_codec_fec, &max_playback_rate);
+ GetOpusConfig(*it, &send_codec, &enable_codec_fec,
+ &opus_max_playback_rate);
}
}
found_send_codec = true;
@@ -2116,15 +2114,16 @@
}
// maxplaybackrate should be set after SetSendCodec.
- if (max_playback_rate > 0) {
+ // If opus_max_playback_rate <= 0, the default maximum playback rate of 48 kHz
+ // will be used.
+ if (opus_max_playback_rate > 0) {
LOG(LS_INFO) << "Attempt to set maximum playback rate to "
- << max_playback_rate
+ << opus_max_playback_rate
<< " Hz on channel "
<< channel;
#ifdef USE_WEBRTC_DEV_BRANCH
- // (max_playback_rate + 1) >> 1 is to obtain ceil(max_playback_rate / 2.0).
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
- channel, max_playback_rate) == -1) {
+ channel, opus_max_playback_rate) == -1) {
LOG(LS_WARNING) << "Could not set maximum playback rate.";
}
#endif
@@ -2133,8 +2132,8 @@
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
- if (send_bw_setting_) {
- SetSendBandwidthInternal(send_bw_bps_);
+ if (send_bitrate_setting_) {
+ SetSendBitrateInternal(send_bitrate_bps_);
}
// Loop through the codecs list again to config the telephone-event/CN codec.
@@ -3187,25 +3186,27 @@
return true;
}
+// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
+// SetMaxSendBitrate() in future.
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChanne::SetSendBandwidth.";
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
- return SetSendBandwidthInternal(bps);
+ return SetSendBitrateInternal(bps);
}
-bool WebRtcVoiceMediaChannel::SetSendBandwidthInternal(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBandwidthInternal.";
+bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
- send_bw_setting_ = true;
- send_bw_bps_ = bps;
+ send_bitrate_setting_ = true;
+ send_bitrate_bps_ = bps;
if (!send_codec_) {
LOG(LS_INFO) << "The send codec has not been set up yet. "
- << "The send bandwidth setting will be applied later.";
+ << "The send bitrate setting will be applied later.";
return true;
}
- // Bandwidth is auto by default.
+ // Bitrate is auto by default.
// TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
// SetMaxSendBandwith(0), the second call removes the previous limit.
if (bps <= 0)
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 8d762d4..f19059b 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -435,7 +435,7 @@
return channel_id == voe_channel();
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
- bool SetSendBandwidthInternal(int bps);
+ bool SetSendBitrateInternal(int bps);
bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
const RtpHeaderExtension* extension);
@@ -453,8 +453,8 @@
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
- bool send_bw_setting_;
- int send_bw_bps_;
+ bool send_bitrate_setting_;
+ int send_bitrate_bps_;
AudioOptions options_;
bool dtmf_allowed_;
bool desired_playout_;
