Allow applications to control audio send bitrate through RtpParameters.
This change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.
The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.
TBR=pthatcher
BUG=
Review URL: https://codereview.webrtc.org/1847353004
Cr-Commit-Position: refs/heads/master@{#12290}
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 21094fd..97e955f 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1081,13 +1081,16 @@
class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioSource::Sink {
public:
- WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
- uint32_t ssrc, const std::string& c_name,
+ WebRtcAudioSendStream(int ch,
+ webrtc::AudioTransport* voe_audio_transport,
+ uint32_t ssrc,
+ const std::string& c_name,
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call)
: voe_audio_transport_(voe_audio_transport),
call_(call),
- config_(nullptr) {
+ config_(nullptr),
+ rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
RTC_DCHECK_GE(ch, 0);
// TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
// RTC_DCHECK(voe_audio_transport);
@@ -1198,6 +1201,15 @@
return config_.voe_channel_id;
}
+ const webrtc::RtpParameters& rtp_parameters() const {
+ return rtp_parameters_;
+ }
+
+ void set_rtp_parameters(const webrtc::RtpParameters& parameters) {
+ RTC_CHECK_EQ(1UL, parameters.encodings.size());
+ rtp_parameters_ = parameters;
+ }
+
private:
void UpdateSendState() {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ -1223,6 +1235,7 @@
// goes away.
AudioSource* source_ = nullptr;
bool send_ = false;
+ webrtc::RtpParameters rtp_parameters_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};
@@ -1359,7 +1372,7 @@
}
}
- if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+ if (!SetSendBitrate(params.max_bandwidth_bps)) {
return false;
}
return SetOptions(params.options);
@@ -1393,6 +1406,51 @@
return true;
}
+webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpParameters(
+ uint32_t ssrc) const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ auto it = send_streams_.find(ssrc);
+ if (it == send_streams_.end()) {
+ LOG(LS_WARNING) << "Attempting to get RTP parameters for stream with ssrc "
+ << ssrc << " which doesn't exist.";
+ return webrtc::RtpParameters();
+ }
+
+ return it->second->rtp_parameters();
+}
+
+bool WebRtcVoiceMediaChannel::SetRtpParameters(
+ uint32_t ssrc,
+ const webrtc::RtpParameters& parameters) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ if (!ValidateRtpParameters(parameters)) {
+ return false;
+ }
+ auto it = send_streams_.find(ssrc);
+ if (it == send_streams_.end()) {
+ LOG(LS_WARNING) << "Attempting to set RTP parameters for stream with ssrc "
+ << ssrc << " which doesn't exist.";
+ return false;
+ }
+
+ if (!SetChannelParameters(it->second->channel(), parameters)) {
+ LOG(LS_WARNING) << "Failed to set RtpParameters.";
+ return false;
+ }
+ it->second->set_rtp_parameters(parameters);
+ return true;
+}
+
+bool WebRtcVoiceMediaChannel::ValidateRtpParameters(
+ const webrtc::RtpParameters& rtp_parameters) {
+ if (rtp_parameters.encodings.size() != 1) {
+ LOG(LS_ERROR)
+ << "Attempted to set RtpParameters without exactly one encoding";
+ return false;
+ }
+ return true;
+}
+
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "Setting voice channel options: "
@@ -1587,7 +1645,7 @@
// Cache the codecs in order to configure the channel created later.
for (const auto& ch : send_streams_) {
- if (!SetSendCodecs(ch.second->channel())) {
+ if (!SetSendCodecs(ch.second->channel(), ch.second->rtp_parameters())) {
return false;
}
}
@@ -1614,7 +1672,9 @@
}
// Apply current codec settings to a single voe::Channel used for sending.
-bool WebRtcVoiceMediaChannel::SetSendCodecs(int channel) {
+bool WebRtcVoiceMediaChannel::SetSendCodecs(
+ int channel,
+ const webrtc::RtpParameters& rtp_parameters) {
// Disable VAD, FEC, and RED unless we know the other side wants them.
engine()->voe()->codec()->SetVADStatus(channel, false);
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
@@ -1682,10 +1742,9 @@
}
}
}
-
- if (send_bitrate_setting_) {
- SetSendBitrateInternal(send_bitrate_bps_);
- }
+ // TODO(solenberg): SetSendBitrate() yields another call to SetSendCodec().
+ // Check if it is possible to fuse with the previous call in this function.
+ SetChannelParameters(channel, rtp_parameters);
// Set the CN payloadtype and the VAD status.
if (send_codec_spec_.cng_payload_type != -1) {
@@ -1880,13 +1939,14 @@
// delete the channel in case failure happens below.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
- send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream(
- channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_)));
+ WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
+ channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_);
+ send_streams_.insert(std::make_pair(ssrc, stream));
// Set the current codecs to be used for the new channel. We need to do this
// after adding the channel to send_channels_, because of how max bitrate is
// currently being configured by SetSendCodec().
- if (HasSendCodec() && !SetSendCodecs(channel)) {
+ if (HasSendCodec() && !SetSendCodecs(channel, stream->rtp_parameters())) {
RemoveSendStream(ssrc);
return false;
}
@@ -2310,18 +2370,34 @@
return true;
}
-// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
-// SetMaxSendBitrate() in future.
-bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
- return SetSendBitrateInternal(bps);
+bool WebRtcVoiceMediaChannel::SetSendBitrate(int bps) {
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrate.";
+ send_bitrate_bps_ = bps;
+
+ for (const auto& kv : send_streams_) {
+ if (!SetChannelParameters(kv.second->channel(),
+ kv.second->rtp_parameters())) {
+ return false;
+ }
+ }
+ return true;
}
-bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
- LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";
+bool WebRtcVoiceMediaChannel::SetChannelParameters(
+ int channel,
+ const webrtc::RtpParameters& parameters) {
+ RTC_CHECK_EQ(1UL, parameters.encodings.size());
+ return SetSendBitrate(
+ channel,
+ MinPositive(send_bitrate_bps_, parameters.encodings[0].max_bitrate_bps));
+}
- send_bitrate_setting_ = true;
- send_bitrate_bps_ = bps;
+bool WebRtcVoiceMediaChannel::SetSendBitrate(int channel, int bps) {
+ // 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)
+ return true;
if (!HasSendCodec()) {
LOG(LS_INFO) << "The send codec has not been set up yet. "
@@ -2329,24 +2405,16 @@
return true;
}
- // 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)
- return true;
-
webrtc::CodecInst codec = send_codec_spec_.codec_inst;
bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
if (is_multi_rate) {
// If codec is multi-rate then just set the bitrate.
codec.rate = bps;
- for (const auto& ch : send_streams_) {
- if (!SetSendCodec(ch.second->channel(), codec)) {
- LOG(LS_INFO) << "Failed to set codec " << codec.plname
- << " to bitrate " << bps << " bps.";
- return false;
- }
+ if (!SetSendCodec(channel, codec)) {
+ LOG(LS_INFO) << "Failed to set codec " << codec.plname << " to bitrate "
+ << bps << " bps.";
+ return false;
}
return true;
} else {
