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-rw-r--r--media/webrtc/webrtcvoiceengine.cc246
1 files changed, 175 insertions, 71 deletions
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 5cf53e4..a524bad 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -52,6 +52,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/video_engine/include/vie_network.h"
#ifdef WIN32
#include <objbase.h> // NOLINT
@@ -119,6 +120,7 @@ static const int kOpusStereoBitrate = 64000;
// Opus bitrate should be in the range between 6000 and 510000.
static const int kOpusMinBitrate = 6000;
static const int kOpusMaxBitrate = 510000;
+
// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
@@ -403,6 +405,7 @@ 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);
}
@@ -429,6 +432,59 @@ static bool IsOpusFecEnabled(const AudioCodec& codec) {
return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
}
+// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
+// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
+static int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
+ int value;
+ if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
+ return value;
+ }
+ return kOpusDefaultMaxPlaybackRate;
+}
+
+static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
+ bool* enable_codec_fec, int* max_playback_rate) {
+ *enable_codec_fec = IsOpusFecEnabled(codec);
+ *max_playback_rate = GetOpusMaxPlaybackRate(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
+ // 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;
+ }
+}
+
void WebRtcVoiceEngine::ConstructCodecs() {
LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
@@ -805,30 +861,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
}
}
- bool experimental_ns;
- if (options.experimental_ns.Get(&experimental_ns)) {
- webrtc::AudioProcessing* audioproc =
- voe_wrapper_->base()->audio_processing();
-#ifdef USE_WEBRTC_DEV_BRANCH
- webrtc::Config config;
- config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(
- experimental_ns));
- audioproc->SetExtraOptions(config);
-#else
- // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
- // returns NULL on audio_processing().
- if (audioproc) {
- if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
- LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
- return false;
- }
- } else {
- LOG(LS_VERBOSE) << "Experimental noise suppression set to "
- << experimental_ns;
- }
-#endif
- }
-
bool highpass_filter;
if (options.highpass_filter.Get(&highpass_filter)) {
LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
@@ -874,20 +906,50 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
StopAecDump();
}
+ webrtc::Config config;
+
+ experimental_aec_.SetFrom(options.experimental_aec);
bool experimental_aec;
- if (options.experimental_aec.Get(&experimental_aec)) {
- LOG(LS_INFO) << "Experimental aec is " << experimental_aec;
- webrtc::AudioProcessing* audioproc =
- voe_wrapper_->base()->audio_processing();
+ if (experimental_aec_.Get(&experimental_aec)) {
+ LOG(LS_INFO) << "Experimental aec is enabled? " << experimental_aec;
+ config.Set<webrtc::DelayCorrection>(
+ new webrtc::DelayCorrection(experimental_aec));
+ }
+
+#ifdef USE_WEBRTC_DEV_BRANCH
+ experimental_ns_.SetFrom(options.experimental_ns);
+ bool experimental_ns;
+ if (experimental_ns_.Get(&experimental_ns)) {
+ LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
+ config.Set<webrtc::ExperimentalNs>(
+ new webrtc::ExperimentalNs(experimental_ns));
+ }
+#endif
+
+ // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
+ // returns NULL on audio_processing().
+ webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
+ if (audioproc) {
+ audioproc->SetExtraOptions(config);
+ }
+
+#ifndef USE_WEBRTC_DEV_BRANCH
+ bool experimental_ns;
+ if (options.experimental_ns.Get(&experimental_ns)) {
+ LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
if (audioproc) {
- webrtc::Config config;
- config.Set<webrtc::DelayCorrection>(
- new webrtc::DelayCorrection(experimental_aec));
- audioproc->SetExtraOptions(config);
+ if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
+ LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
+ return false;
+ }
+ } else {
+ LOG(LS_VERBOSE) << "Experimental noise suppression set to "
+ << experimental_ns;
}
}
+#endif
uint32 recording_sample_rate;
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
@@ -1759,6 +1821,8 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
typing_noise_detected_(false),
desired_send_(SEND_NOTHING),
send_(SEND_NOTHING),
+ shared_bwe_vie_(NULL),
+ shared_bwe_vie_channel_(-1),
default_receive_ssrc_(0) {
engine->RegisterChannel(this);
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
@@ -1770,6 +1834,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
<< voe_channel();
+ SetupSharedBandwidthEstimation(NULL, -1);
// Remove any remaining send streams, the default channel will be deleted
// later.
@@ -1870,6 +1935,12 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
}
}
+ // Force update of Video Engine BWE forwarding to reflect experiment setting.
+ if (!SetupSharedBandwidthEstimation(shared_bwe_vie_,
+ shared_bwe_vie_channel_)) {
+ return false;
+ }
+
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
return true;
@@ -1976,6 +2047,10 @@ 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;
+
// Set send codec (the first non-telephone-event/CN codec)
for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
it != codecs.end(); ++it) {
@@ -1992,40 +2067,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
continue;
}
- // 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
- // appropriate default value for mono or stereo Opus.
- if (IsOpus(*it)) {
- if (IsOpusStereoEnabled(*it)) {
- voe_codec.channels = 2;
- if (!IsValidOpusBitrate(it->bitrate)) {
- if (it->bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << it->bitrate
- << ") with default opus stereo bitrate: "
- << kOpusStereoBitrate;
- }
- voe_codec.rate = kOpusStereoBitrate;
- }
- } else {
- voe_codec.channels = 1;
- if (!IsValidOpusBitrate(it->bitrate)) {
- if (it->bitrate != 0) {
- LOG(LS_WARNING) << "Overrides the invalid supplied bitrate("
- << it->bitrate
- << ") with default opus mono bitrate: "
- << kOpusMonoBitrate;
- }
- voe_codec.rate = kOpusMonoBitrate;
- }
- }
- int bitrate_from_params = GetOpusBitrateFromParams(*it);
- if (bitrate_from_params != 0) {
- voe_codec.rate = bitrate_from_params;
- }
- }
// 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.
@@ -2055,8 +2096,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
} else {
send_codec = voe_codec;
nack_enabled = IsNackEnabled(*it);
- // For Opus as the send codec, we enable inband FEC if requested.
- enable_codec_fec = IsOpus(*it) && IsOpusFecEnabled(*it);
+ // 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);
+ }
}
found_send_codec = true;
break;
@@ -2090,6 +2134,21 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
#endif // USE_WEBRTC_DEV_BRANCH
}
+ // maxplaybackrate should be set after SetSendCodec.
+ if (max_playback_rate > 0) {
+ LOG(LS_INFO) << "Attempt to set maximum playback rate to "
+ << 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) {
+ LOG(LS_WARNING) << "Could not set maximum playback rate.";
+ }
+#endif
+ }
+
// Always update the |send_codec_| to the currently set send codec.
send_codec_.reset(new webrtc::CodecInst(send_codec));
@@ -2531,8 +2590,8 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
}
if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
- LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
- return false;
+ LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
+ return false;
}
// Set the current codecs to be used for the new channel.
@@ -2604,6 +2663,9 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
receive_channels_.insert(std::make_pair(
default_receive_ssrc_,
new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport)));
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
return SetPlayout(voe_channel(), playout_);
}
@@ -2691,6 +2753,11 @@ bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
return false;
}
+ // Set up channel to be able to forward incoming packets to video engine BWE.
+ if (!SetupSharedBweOnChannel(channel)) {
+ return false;
+ }
+
return SetPlayout(channel, playout_);
}
@@ -3060,7 +3127,8 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
engine()->voe()->network()->ReceivedRTPPacket(
which_channel,
packet->data(),
- static_cast<unsigned int>(packet->length()));
+ static_cast<unsigned int>(packet->length()),
+ webrtc::PacketTime(packet_time.timestamp, packet_time.not_before));
}
void WebRtcVoiceMediaChannel::OnRtcpReceived(
@@ -3454,6 +3522,23 @@ int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) {
return -1;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBandwidthEstimation(
+ webrtc::VideoEngine* vie, int vie_channel) {
+ shared_bwe_vie_ = vie;
+ shared_bwe_vie_channel_ = vie_channel;
+
+ if (!SetupSharedBweOnChannel(voe_channel())) {
+ return false;
+ }
+ for (ChannelMap::iterator it = receive_channels_.begin();
+ it != receive_channels_.end(); ++it) {
+ if (!SetupSharedBweOnChannel(it->second->channel())) {
+ return false;
+ }
+ }
+ return true;
+}
+
bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
// Get the RED encodings from the parameter with no name. This may
@@ -3603,6 +3688,25 @@ bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
return true;
}
+bool WebRtcVoiceMediaChannel::SetupSharedBweOnChannel(int voe_channel) {
+ webrtc::ViENetwork* vie_network = NULL;
+ int vie_channel = -1;
+ if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false) &&
+ shared_bwe_vie_ != NULL && shared_bwe_vie_channel_ != -1) {
+ vie_network = webrtc::ViENetwork::GetInterface(shared_bwe_vie_);
+ vie_channel = shared_bwe_vie_channel_;
+ }
+ if (engine()->voe()->rtp()->SetVideoEngineBWETarget(voe_channel, vie_network,
+ vie_channel) == -1) {
+ LOG_RTCERR3(SetVideoEngineBWETarget, voe_channel, vie_network, vie_channel);
+ if (vie_network != NULL) {
+ // Don't fail if we're tearing down.
+ return false;
+ }
+ }
+ return true;
+}
+
int WebRtcSoundclipStream::Read(void *buf, int len) {
size_t res = 0;
mem_.Read(buf, len, &res, NULL);
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-26 00:38:30 +0000