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Diffstat (limited to 'media/webrtc/webrtcvoiceengine.cc')
-rw-r--r--media/webrtc/webrtcvoiceengine.cc143
1 files changed, 72 insertions, 71 deletions
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index f9e7878..95e16e4 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -110,13 +110,26 @@ static const int kDefaultAudioDeviceId = 0;
static const char kIsacCodecName[] = "ISAC";
static const char kL16CodecName[] = "L16";
-// Codec parameters for Opus.
-static const int kOpusMonoBitrate = 32000;
+
// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
static const int kNackMaxPackets = 250;
-static const int kOpusStereoBitrate = 64000;
+
+// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03
+
+// Recommended bitrates:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation applies the following values to mono signals,
+// and multiplies them by 2 for stereo.
+static const int kOpusBitrateNb = 12000;
+static const int kOpusBitrateWb = 20000;
+static const int kOpusBitrateFb = 32000;
+
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
@@ -405,22 +418,37 @@ static bool IsOpusStereoEnabled(const AudioCodec& codec) {
return codec.GetParam(kCodecParamStereo, &value) && value == 1;
}
-// TODO(minyue): Clamp bitrate when invalid.
-static bool IsValidOpusBitrate(int bitrate) {
- return (bitrate >= kOpusMinBitrate && bitrate <= kOpusMaxBitrate);
-}
-
-// Returns 0 if params[kCodecParamMaxAverageBitrate] is not defined or invalid.
-// Returns the value of params[kCodecParamMaxAverageBitrate] otherwise.
-static int GetOpusBitrateFromParams(const AudioCodec& codec) {
+// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
+// otherwise. If the value (either from params or codec.bitrate) <=0, use the
+// default configuration. If the value is beyond feasible bit rate of Opus,
+// clamp it. Returns the Opus bit rate for operation.
+static int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
int bitrate = 0;
+ bool use_param = true;
if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
- return 0;
- }
- if (!IsValidOpusBitrate(bitrate)) {
- LOG(LS_WARNING) << "Codec parameter \"maxaveragebitrate\" has an "
- << "invalid value: " << bitrate;
- return 0;
+ bitrate = codec.bitrate;
+ use_param = false;
+ }
+ if (bitrate <= 0) {
+ if (max_playback_rate <= 8000) {
+ bitrate = kOpusBitrateNb;
+ } else if (max_playback_rate <= 16000) {
+ bitrate = kOpusBitrateWb;
+ } else {
+ bitrate = kOpusBitrateFb;
+ }
+
+ if (IsOpusStereoEnabled(codec)) {
+ bitrate *= 2;
+ }
+ } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
+ bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
+ std::string rate_source =
+ use_param ? "Codec parameter \"maxaveragebitrate\"" :
+ "Supplied Opus bitrate";
+ LOG(LS_WARNING) << rate_source
+ << " is invalid and is replaced by: "
+ << bitrate;
}
return bitrate;
}
@@ -450,39 +478,11 @@ static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
// If OPUS, change what we send according to the "stereo" codec
// parameter, and not the "channels" parameter. We set
// voe_codec.channels to 2 if "stereo=1" and 1 otherwise. If
- // the bitrate is not specified, i.e. is zero, we set it to the
+ // the bitrate is not specified, i.e. is <= zero, we set it to the
// appropriate default value for mono or stereo Opus.
- // TODO(minyue): The determination of bit rate might take the maximum playback
- // rate into account.
-
- if (IsOpusStereoEnabled(codec)) {
- voe_codec->channels = 2;
- if (!IsValidOpusBitrate(codec.bitrate)) {
- if (codec.bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << codec.bitrate
- << ") with default opus stereo bitrate: "
- << kOpusStereoBitrate;
- }
- voe_codec->rate = kOpusStereoBitrate;
- }
- } else {
- voe_codec->channels = 1;
- if (!IsValidOpusBitrate(codec.bitrate)) {
- if (codec.bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << codec.bitrate
- << ") with default opus mono bitrate: "
- << kOpusMonoBitrate;
- }
- voe_codec->rate = kOpusMonoBitrate;
- }
- }
- int bitrate_from_params = GetOpusBitrateFromParams(codec);
- if (bitrate_from_params != 0) {
- voe_codec->rate = bitrate_from_params;
- }
+ voe_codec->channels = IsOpusStereoEnabled(codec) ? 2 : 1;
+ voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
void WebRtcVoiceEngine::ConstructCodecs() {
@@ -514,7 +514,7 @@ void WebRtcVoiceEngine::ConstructCodecs() {
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)) {
@@ -1246,7 +1246,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
// 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;
@@ -1811,8 +1811,8 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
: 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),
@@ -2047,9 +2047,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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();
@@ -2067,7 +2065,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
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
@@ -2099,7 +2096,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// 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;
@@ -2135,15 +2133,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
}
// 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
@@ -2152,8 +2151,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// 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.
@@ -3206,25 +3205,27 @@ bool WebRtcVoiceMediaChannel::MuteStream(uint32 ssrc, bool muted) {
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)
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-23 00:57:36 +0000